[Aug 18 10:31:30] Asterisk 16.20.0 built by root @ GolosBetaAsterisk-01 on a x86_64 running Linux on 2021-08-17 13:06:28 UTC [Aug 18 10:31:30] VERBOSE[11659] logger.c: Asterisk Queue Logger restarted [Aug 18 10:31:35] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:31:35] VERBOSE[11721] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:31:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:31:48] VERBOSE[11993] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:31:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:31:48] VERBOSE[12002] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:31:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:31:48] VERBOSE[12012] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:31:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:31:48] VERBOSE[12021] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:00] Asterisk 16.20.0 built by root @ GolosBetaAsterisk-01 on a x86_64 running Linux on 2021-08-17 13:06:28 UTC [Aug 18 10:32:00] VERBOSE[11659] loader.c: Reloading module 'logger' (Logger) [Aug 18 10:32:00] DEBUG[11659] config.c: Parsing /etc/asterisk/logger.conf [Aug 18 10:32:00] VERBOSE[11659] logger.c: Asterisk Queue Logger restarted [Aug 18 10:32:03] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:03] VERBOSE[12064] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:19] DEBUG[20564] res_pjsip_registrar.c: Woke up at 1629282739 Interval: 30 [Aug 18 10:32:19] DEBUG[20564] res_pjsip_registrar.c: Expiring 0 contacts [Aug 18 10:32:34] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:34] VERBOSE[12260] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:47] VERBOSE[12522] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:47] VERBOSE[12528] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:47] VERBOSE[12535] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:32:48] VERBOSE[12543] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:32:49] DEBUG[20564] res_pjsip_registrar.c: Woke up at 1629282769 Interval: 30 [Aug 18 10:32:49] DEBUG[20564] res_pjsip_registrar.c: Expiring 0 contacts [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Allocating new SIP dialog for 2cc29e9f7619e9e029a37f3974c61d86@127.0.1.1:5060 - OPTIONS (No RTP) [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:32:50] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: SIP call-id changed from '2cc29e9f7619e9e029a37f3974c61d86@127.0.1.1:5060' to '7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060' [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Initializing initreq for method OPTIONS - callid 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 33]: OPTIONS sip:178.62.121.41 SIP/2.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4851c933 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as34f8847c [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 23]: To: [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 42]: Contact: [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 60]: Call-ID: 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:32:50 GMT [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: OPTIONS sip:178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4851c933 Max-Forwards: 70 From: "asterisk" ;tag=as34f8847c To: Contact: Call-ID: 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:32:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4851c933;received=159.65.48.104 From: "asterisk" ;tag=as34f8847c To: ;tag=as6c743673 Call-ID: 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4851c933;received=159.65.48.104 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as34f8847c [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 38]: To: ;tag=as6c743673 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 (Checking To) --From tag as34f8847c --To-tag as6c743673 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7e19718d4c3e419868f5b4b37c796ce6@159.65.48.104:5060' Method: OPTIONS [Aug 18 10:32:50] NOTICE[20585] chan_sip.c: -- Re-registration for zvonobot@178.62.121.41 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Allocating new SIP dialog for 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 - REGISTER (No RTP) [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:32:50] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: >>> Re-using Auth data for zvonobot@178.62.121.41 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Initializing initreq for method REGISTER - callid 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 34]: REGISTER sip:178.62.121.41 SIP/2.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40c92b5f [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 49]: From: ;tag=as0e88e9e0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 32]: To: [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 51]: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 18]: CSeq: 254 REGISTER [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [162]: Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="37b3e635", response="fd806141b68f2144a81df608e68704bd" [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 12]: Expires: 120 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 11 [ 42]: Contact: [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: REGISTER 12 headers, 0 lines [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: REGISTER attempt 1 to zvonobot@178.62.121.41 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: REGISTER sip:178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40c92b5f Max-Forwards: 70 From: ;tag=as0e88e9e0 To: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 CSeq: 254 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="37b3e635", response="fd806141b68f2144a81df608e68704bd" Expires: 120 Contact: Content-Length: 0 --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #12 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Scheduled a registration timeout for 178.62.121.41 id #4 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40c92b5f;received=159.65.48.104 From: ;tag=as0e88e9e0 To: ;tag=as5f484490 Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 CSeq: 254 REGISTER Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d10b840" Content-Length: 0 <-------------> [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40c92b5f;received=159.65.48.104 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 49]: From: ;tag=as0e88e9e0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 47]: To: ;tag=as5f484490 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 51]: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 18]: CSeq: 254 REGISTER [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d10b840" [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 (Checking To) --From tag as0e88e9e0 --To-tag as5f484490 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #12 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1' of Request 254: Match Found [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Responding to challenge, registration to domain/host name 178.62.121.41 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:32:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Initializing already initialized SIP dialog 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 (presumably reinvite) [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 34]: REGISTER sip:178.62.121.41 SIP/2.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK13787102 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 49]: From: ;tag=as0e88e9e0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 32]: To: [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 51]: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 18]: CSeq: 255 REGISTER [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [162]: Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1d10b840", response="f8924d51befdaa09a09eff211a53a875" [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 12]: Expires: 120 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 11 [ 42]: Contact: [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: REGISTER 12 headers, 0 lines [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: REGISTER attempt 2 to zvonobot@178.62.121.41 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: REGISTER sip:178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK13787102 Max-Forwards: 70 From: ;tag=as0e88e9e0 To: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 CSeq: 255 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1d10b840", response="f8924d51befdaa09a09eff211a53a875" Expires: 120 Contact: Content-Length: 0 --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK13787102;received=159.65.48.104 From: ;tag=as0e88e9e0 To: ;tag=as5f484490 Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 CSeq: 255 REGISTER Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 18 Aug 2021 10:32:50 GMT Content-Length: 0 <-------------> [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK13787102;received=159.65.48.104 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 2 [ 49]: From: ;tag=as0e88e9e0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 3 [ 47]: To: ;tag=as5f484490 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 4 [ 51]: Call-ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 5 [ 18]: CSeq: 255 REGISTER [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 9 [ 12]: Expires: 120 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 10 [ 54]: Contact: ;expires=120 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 11 [ 35]: Date: Wed, 18 Aug 2021 10:32:50 GMT [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: --- (13 headers 0 lines) --- [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 (Checking To) --From tag as0e88e9e0 --To-tag as5f484490 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1' of Request 255: Match Found [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Registration successful [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Cancelling timeout 4 [Aug 18 10:32:50] NOTICE[20585] chan_sip.c: Outbound Registration: Expiry for 178.62.121.41 is 120 sec (Scheduling reregistration in 105 s) [Aug 18 10:32:50] DEBUG[20585] chan_sip.c: Destroying SIP dialog 0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1 [Aug 18 10:32:50] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '0c2b3bfb511d3af148d6c35c367fa186@127.0.1.1' Method: REGISTER [Aug 18 10:33:03] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:03] VERBOSE[12553] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:19] DEBUG[20564] res_pjsip_registrar.c: Woke up at 1629282799 Interval: 30 [Aug 18 10:33:19] DEBUG[20564] res_pjsip_registrar.c: Expiring 0 contacts [Aug 18 10:33:33] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:33] VERBOSE[12716] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:41] DEBUG[12864] http.c: HTTP opening session. Top level [Aug 18 10:33:41] DEBUG[12864] http.c: HTTP Request URI is /ari/channels/212964?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117076&callerId=74950493843 [Aug 18 10:33:41] DEBUG[12864] http.c: match request [ari/channels/212964] with handler [httpstatus] len 10 [Aug 18 10:33:41] DEBUG[12864] http.c: match request [ari/channels/212964] with handler [phoneprov] len 9 [Aug 18 10:33:41] DEBUG[12864] http.c: match request [ari/channels/212964] with handler [ari] len 3 [Aug 18 10:33:41] DEBUG[12864] http.c: Match made with [ari] [Aug 18 10:33:41] DEBUG[12864] http.c: HTTP consuming request body [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Finding handler for channels/212964 [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Finding handler for channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Finding handler for 212964 [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking channels create: Didn't match 212964 [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:41] DEBUG[12864] res_ari.c: Checking channels externalMedia: Didn't match 212964 [Aug 18 10:33:41] DEBUG[12864] res_ari.c: No explicit handler found for 212964. Using wildcard channelId. [Aug 18 10:33:41] DEBUG[12864] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: Allocating new SIP dialog for 67f028080a07763a2512a3722f204234@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12864] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0001f70' [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP allocated port 15726 [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE creating session 0.0.0.0:15726 (15726) [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE create [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE add system candidates [Aug 18 10:33:42] DEBUG[12864] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12864] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE add candidate: 159.65.48.104:15726, 2130706431 [Aug 18 10:33:42] DEBUG[12864] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12864] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE add candidate: 10.131.0.10:15726, 2130706431 [Aug 18 10:33:42] DEBUG[12864] rtp_engine.c: RTP instance '0x7f0cb0001f70' is setup and ready to go [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE stopped [Aug 18 10:33:42] DEBUG[12864] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12864] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12864] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12864] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12864] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12864] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: SIP call-id changed from '67f028080a07763a2512a3722f204234@127.0.1.1:5060' to '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12864] stasis.c: Creating topic. name: channel:212964, detail: [Aug 18 10:33:42] DEBUG[12864] stasis.c: Topic 'channel:212964': 0x7f0cb00142d0 created [Aug 18 10:33:42] DEBUG[12864] stasis.c: Creating topic. name: cache:7/channel:212964, detail: [Aug 18 10:33:42] DEBUG[12864] stasis.c: Topic 'cache:7/channel:212964': 0x7f0cb0014500 created [Aug 18 10:33:42] DEBUG[12864] channel.c: Channel 0x7f0cb0014a50 'SIP/zvonobot-00000000' allocated [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12864] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12864] res_stasis.c: calls_0: Subscribing to 212964 [Aug 18 10:33:42] DEBUG[12864] stasis/app.c: Channel '212964' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Outgoing Call for 79821117076 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12864] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12865] chan_sip.c: Audio is at 15726 [Aug 18 10:33:42] VERBOSE[12865] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12865] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] DEBUG[12864] http.c: HTTP closing session. Top level [Aug 18 10:33:42] VERBOSE[12865] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Initializing initreq for method INVITE - callid 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117076@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400951a3 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 3 [ 52]: From: ;tag=as39d9ed01 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 6 [ 60]: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:41 GMT [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12865] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117076@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400951a3 Max-Forwards: 70 From: ;tag=as39d9ed01 To: Contact: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 124370842 124370842 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:42] DEBUG[12865] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12865] dial.c: Called zvonobot/79821117076 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400951a3;received=159.65.48.104 From: ;tag=as39d9ed01 To: ;tag=as36c3c077 Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11c410aa" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400951a3;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39d9ed01 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as36c3c077 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11c410aa" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 (Checking To) --From tag as39d9ed01 --To-tag as36c3c077 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117076@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400951a3 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as36c3c077 Contact: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 15726 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117076@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK234c8021 Max-Forwards: 70 From: ;tag=as39d9ed01 To: Contact: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41", nonce="11c410aa", response="2a1d5ee621d6ef0c29a27231a0dac120" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 124370842 124370843 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK234c8021;received=159.65.48.104 From: ;tag=as39d9ed01 To: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK234c8021;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39d9ed01 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 (Checking To) --From tag as39d9ed01 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12868] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12868] http.c: HTTP Request URI is /ari/channels/212965?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117075&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12868] http.c: match request [ari/channels/212965] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12868] http.c: match request [ari/channels/212965] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12868] http.c: match request [ari/channels/212965] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12868] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12868] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Finding handler for channels/212965 [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Finding handler for 212965 [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking channels create: Didn't match 212965 [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12868] res_ari.c: Checking channels externalMedia: Didn't match 212965 [Aug 18 10:33:42] DEBUG[12868] res_ari.c: No explicit handler found for 212965. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: Allocating new SIP dialog for 54702c2a2aa4ddea43c87b32413244bf@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12868] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4008d90' [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP allocated port 19848 [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE creating session 0.0.0.0:19848 (19848) [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE create [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE add system candidates [Aug 18 10:33:42] DEBUG[12868] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12868] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE add candidate: 159.65.48.104:19848, 2130706431 [Aug 18 10:33:42] DEBUG[12868] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12868] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE add candidate: 10.131.0.10:19848, 2130706431 [Aug 18 10:33:42] DEBUG[12868] rtp_engine.c: RTP instance '0x7f0cb4008d90' is setup and ready to go [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE stopped [Aug 18 10:33:42] DEBUG[12868] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12868] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12868] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12868] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12868] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12868] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: SIP call-id changed from '54702c2a2aa4ddea43c87b32413244bf@127.0.1.1:5060' to '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12868] stasis.c: Creating topic. name: channel:212965, detail: [Aug 18 10:33:42] DEBUG[12868] stasis.c: Topic 'channel:212965': 0x7f0cb4011f50 created [Aug 18 10:33:42] DEBUG[12868] stasis.c: Creating topic. name: cache:8/channel:212965, detail: [Aug 18 10:33:42] DEBUG[12868] stasis.c: Topic 'cache:8/channel:212965': 0x7f0cb4011b20 created [Aug 18 10:33:42] DEBUG[12868] channel.c: Channel 0x7f0cb40101d0 'SIP/zvonobot-00000001' allocated [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12868] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12868] res_stasis.c: calls_0: Subscribing to 212965 [Aug 18 10:33:42] DEBUG[12868] stasis/app.c: Channel '212965' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Outgoing Call for 79821117075 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12868] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12869] chan_sip.c: Audio is at 19848 [Aug 18 10:33:42] VERBOSE[12869] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12869] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12869] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Initializing initreq for method INVITE - callid 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117075@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7accfa9a [Aug 18 10:33:42] DEBUG[12868] http.c: HTTP closing session. Top level [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 3 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 6 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12869] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117075@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7accfa9a Max-Forwards: 70 From: ;tag=as7d114a77 To: Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78981942 78981942 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19848 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:42] DEBUG[12869] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12869] dial.c: Called zvonobot/79821117075 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7accfa9a;received=159.65.48.104 From: ;tag=as7d114a77 To: ;tag=as5859e0da Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e661060" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7accfa9a;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5859e0da [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e661060" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 (Checking To) --From tag as7d114a77 --To-tag as5859e0da [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117075@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7accfa9a Max-Forwards: 70 From: ;tag=as7d114a77 To: ;tag=as5859e0da Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 19848 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117075@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f Max-Forwards: 70 From: ;tag=as7d114a77 To: Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117075@178.62.121.41", nonce="1e661060", response="6c2a1b7df9878a6d9ded99b1b8d6c984" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78981942 78981943 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19848 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 From: ;tag=as7d114a77 To: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 (Checking To) --From tag as7d114a77 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12872] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12872] http.c: HTTP Request URI is /ari/channels/212966?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117074&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12872] http.c: match request [ari/channels/212966] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12872] http.c: match request [ari/channels/212966] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12872] http.c: match request [ari/channels/212966] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12872] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12872] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Finding handler for channels/212966 [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Finding handler for 212966 [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking channels create: Didn't match 212966 [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12872] res_ari.c: Checking channels externalMedia: Didn't match 212966 [Aug 18 10:33:42] DEBUG[12872] res_ari.c: No explicit handler found for 212966. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: Allocating new SIP dialog for 26beb2e567b80d71449ea8d20862cbfc@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12872] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1000e000' [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) RTP allocated port 17878 [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE creating session 0.0.0.0:17878 (17878) [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE create [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE add system candidates [Aug 18 10:33:42] DEBUG[12872] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12872] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE add candidate: 159.65.48.104:17878, 2130706431 [Aug 18 10:33:42] DEBUG[12872] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12872] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE add candidate: 10.131.0.10:17878, 2130706431 [Aug 18 10:33:42] DEBUG[12872] rtp_engine.c: RTP instance '0x7f0c1000e000' is setup and ready to go [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) ICE stopped [Aug 18 10:33:42] DEBUG[12872] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12872] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12872] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12872] res_rtp_asterisk.c: (0x7f0c1000e000) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12872] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12872] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: SIP call-id changed from '26beb2e567b80d71449ea8d20862cbfc@127.0.1.1:5060' to '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12872] stasis.c: Creating topic. name: channel:212966, detail: [Aug 18 10:33:42] DEBUG[12872] stasis.c: Topic 'channel:212966': 0x7f0c10015e40 created [Aug 18 10:33:42] DEBUG[12872] stasis.c: Creating topic. name: cache:9/channel:212966, detail: [Aug 18 10:33:42] DEBUG[12872] stasis.c: Topic 'cache:9/channel:212966': 0x7f0c100174a0 created [Aug 18 10:33:42] DEBUG[12872] channel.c: Channel 0x7f0c100160e0 'SIP/zvonobot-00000002' allocated [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12872] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12872] res_stasis.c: calls_0: Subscribing to 212966 [Aug 18 10:33:42] DEBUG[12872] stasis/app.c: Channel '212966' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12872] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12872] http.c: HTTP closing session. Top level [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Outgoing Call for 79821117074 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12874] chan_sip.c: Audio is at 17878 [Aug 18 10:33:42] VERBOSE[12874] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12874] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12874] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Initializing initreq for method INVITE - callid 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117074@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1487f743 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 3 [ 52]: From: ;tag=as08e169d8 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 6 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12874] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117074@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1487f743 Max-Forwards: 70 From: ;tag=as08e169d8 To: Contact: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 929349489 929349489 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17878 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:42] DEBUG[12874] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1487f743;received=159.65.48.104 From: ;tag=as08e169d8 To: ;tag=as1c118663 Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="560ef8b8" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1487f743;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08e169d8 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1c118663 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="560ef8b8" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 (Checking To) --From tag as08e169d8 --To-tag as1c118663 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117074@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1487f743 Max-Forwards: 70 From: ;tag=as08e169d8 To: ;tag=as1c118663 Contact: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 17878 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117074@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882 Max-Forwards: 70 From: ;tag=as08e169d8 To: Contact: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117074@178.62.121.41", nonce="560ef8b8", response="9b7fb90c9b5d5ecf138eea9686267572" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 929349489 929349490 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17878 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12874] dial.c: Called zvonobot/79821117074 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 From: ;tag=as08e169d8 To: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08e169d8 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 (Checking To) --From tag as08e169d8 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12877] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12877] http.c: HTTP Request URI is /ari/channels/212968?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117072&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12877] http.c: match request [ari/channels/212968] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12877] http.c: match request [ari/channels/212968] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12877] http.c: match request [ari/channels/212968] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12877] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12877] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Finding handler for channels/212968 [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Finding handler for 212968 [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking channels create: Didn't match 212968 [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12877] res_ari.c: Checking channels externalMedia: Didn't match 212968 [Aug 18 10:33:42] DEBUG[12877] res_ari.c: No explicit handler found for 212968. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: Allocating new SIP dialog for 7e3f899d27f755e45021a0e36a609f84@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12877] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c00f0c0' [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) RTP allocated port 12550 [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE creating session 0.0.0.0:12550 (12550) [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE create [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE add system candidates [Aug 18 10:33:42] DEBUG[12877] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12877] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE add candidate: 159.65.48.104:12550, 2130706431 [Aug 18 10:33:42] DEBUG[12877] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12877] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE add candidate: 10.131.0.10:12550, 2130706431 [Aug 18 10:33:42] DEBUG[12877] rtp_engine.c: RTP instance '0x7f0c1c00f0c0' is setup and ready to go [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE stopped [Aug 18 10:33:42] DEBUG[12877] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12877] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12877] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12877] res_rtp_asterisk.c: (0x7f0c1c00f0c0) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12877] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12877] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: SIP call-id changed from '7e3f899d27f755e45021a0e36a609f84@127.0.1.1:5060' to '72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12877] stasis.c: Creating topic. name: channel:212968, detail: [Aug 18 10:33:42] DEBUG[12877] stasis.c: Topic 'channel:212968': 0x7f0c1c018590 created [Aug 18 10:33:42] DEBUG[12877] stasis.c: Creating topic. name: cache:10/channel:212968, detail: [Aug 18 10:33:42] DEBUG[12877] stasis.c: Topic 'cache:10/channel:212968': 0x7f0c1c07cfc0 created [Aug 18 10:33:42] DEBUG[12877] channel.c: Channel 0x7f0c1c016880 'SIP/zvonobot-00000003' allocated [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12877] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12877] res_stasis.c: calls_0: Subscribing to 212968 [Aug 18 10:33:42] DEBUG[12877] stasis/app.c: Channel '212968' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Outgoing Call for 79821117072 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12877] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12879] chan_sip.c: Audio is at 12550 [Aug 18 10:33:42] VERBOSE[12879] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] DEBUG[12877] http.c: HTTP closing session. Top level [Aug 18 10:33:42] VERBOSE[12879] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12879] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Initializing initreq for method INVITE - callid 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117072@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK23ff052b [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 3 [ 52]: From: ;tag=as6af53e10 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 6 [ 60]: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12879] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117072@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK23ff052b Max-Forwards: 70 From: ;tag=as6af53e10 To: Contact: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 667341273 667341273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12550 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:42] DEBUG[12879] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12879] dial.c: Called zvonobot/79821117072 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK23ff052b;received=159.65.48.104 From: ;tag=as6af53e10 To: ;tag=as60188ade Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26fb2619" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK23ff052b;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6af53e10 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as60188ade [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26fb2619" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 (Checking To) --From tag as6af53e10 --To-tag as60188ade [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117072@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK23ff052b Max-Forwards: 70 From: ;tag=as6af53e10 To: ;tag=as60188ade Contact: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 12550 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117072@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517 Max-Forwards: 70 From: ;tag=as6af53e10 To: Contact: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117072@178.62.121.41", nonce="26fb2619", response="98a5b6c620ee5c7a82d6f379ab47d705" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 667341273 667341274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12550 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517;received=159.65.48.104 From: ;tag=as6af53e10 To: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6af53e10 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 (Checking To) --From tag as6af53e10 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12883] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12883] http.c: HTTP Request URI is /ari/channels/212967?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117073&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12883] http.c: match request [ari/channels/212967] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12883] http.c: match request [ari/channels/212967] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12883] http.c: match request [ari/channels/212967] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12883] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12883] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Finding handler for channels/212967 [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Finding handler for 212967 [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking channels create: Didn't match 212967 [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12883] res_ari.c: Checking channels externalMedia: Didn't match 212967 [Aug 18 10:33:42] DEBUG[12883] res_ari.c: No explicit handler found for 212967. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: Allocating new SIP dialog for 4fd392b7286ced8129b265b401311392@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12883] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2400b7d0' [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTP allocated port 18520 [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE creating session 0.0.0.0:18520 (18520) [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE create [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE add system candidates [Aug 18 10:33:42] DEBUG[12883] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12883] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE add candidate: 159.65.48.104:18520, 2130706431 [Aug 18 10:33:42] DEBUG[12883] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12883] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE add candidate: 10.131.0.10:18520, 2130706431 [Aug 18 10:33:42] DEBUG[12883] rtp_engine.c: RTP instance '0x7f0c2400b7d0' is setup and ready to go [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE stopped [Aug 18 10:33:42] DEBUG[12883] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12883] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12883] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12883] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12883] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12883] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: SIP call-id changed from '4fd392b7286ced8129b265b401311392@127.0.1.1:5060' to '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12883] stasis.c: Creating topic. name: channel:212967, detail: [Aug 18 10:33:42] DEBUG[12883] stasis.c: Topic 'channel:212967': 0x7f0c24078190 created [Aug 18 10:33:42] DEBUG[12883] stasis.c: Creating topic. name: cache:11/channel:212967, detail: [Aug 18 10:33:42] DEBUG[12883] stasis.c: Topic 'cache:11/channel:212967': 0x7f0c24078350 created [Aug 18 10:33:42] DEBUG[12883] channel.c: Channel 0x7f0c24011df0 'SIP/zvonobot-00000004' allocated [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12883] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12883] res_stasis.c: calls_0: Subscribing to 212967 [Aug 18 10:33:42] DEBUG[12883] stasis/app.c: Channel '212967' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Outgoing Call for 79821117073 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12883] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12888] chan_sip.c: Audio is at 18520 [Aug 18 10:33:42] VERBOSE[12888] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12888] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12888] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Initializing initreq for method INVITE - callid 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12883] http.c: HTTP closing session. Top level [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117073@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ceeb0fc [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 3 [ 52]: From: ;tag=as28933467 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 6 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12888] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117073@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ceeb0fc Max-Forwards: 70 From: ;tag=as28933467 To: Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20384970 20384970 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18520 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:42] DEBUG[12888] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12888] dial.c: Called zvonobot/79821117073 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ceeb0fc;received=159.65.48.104 From: ;tag=as28933467 To: ;tag=as4773ebdc Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="335d9687" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ceeb0fc;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28933467 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4773ebdc [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="335d9687" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking To) --From tag as28933467 --To-tag as4773ebdc [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117073@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ceeb0fc Max-Forwards: 70 From: ;tag=as28933467 To: ;tag=as4773ebdc Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 18520 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117073@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c Max-Forwards: 70 From: ;tag=as28933467 To: Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117073@178.62.121.41", nonce="335d9687", response="587c458f5900e061c9e853a341fad60f" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20384970 20384971 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18520 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 From: ;tag=as28933467 To: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28933467 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking To) --From tag as28933467 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12890] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12890] http.c: HTTP Request URI is /ari/channels/212970?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117070&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12890] http.c: match request [ari/channels/212970] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12890] http.c: match request [ari/channels/212970] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12890] http.c: match request [ari/channels/212970] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12890] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12890] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Finding handler for channels/212970 [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Finding handler for 212970 [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking channels create: Didn't match 212970 [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12890] res_ari.c: Checking channels externalMedia: Didn't match 212970 [Aug 18 10:33:42] DEBUG[12890] res_ari.c: No explicit handler found for 212970. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: Allocating new SIP dialog for 54355470719ef9940708064b09812bed@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12890] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c00bbb0' [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) RTP allocated port 12972 [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE creating session 0.0.0.0:12972 (12972) [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE create [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE add system candidates [Aug 18 10:33:42] DEBUG[12890] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12890] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE add candidate: 159.65.48.104:12972, 2130706431 [Aug 18 10:33:42] DEBUG[12890] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12890] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE add candidate: 10.131.0.10:12972, 2130706431 [Aug 18 10:33:42] DEBUG[12890] rtp_engine.c: RTP instance '0x7f0c2c00bbb0' is setup and ready to go [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE stopped [Aug 18 10:33:42] DEBUG[12890] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12890] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12890] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12890] res_rtp_asterisk.c: (0x7f0c2c00bbb0) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12890] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12890] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: SIP call-id changed from '54355470719ef9940708064b09812bed@127.0.1.1:5060' to '36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12890] stasis.c: Creating topic. name: channel:212970, detail: [Aug 18 10:33:42] DEBUG[12890] stasis.c: Topic 'channel:212970': 0x7f0c2c012760 created [Aug 18 10:33:42] DEBUG[12890] stasis.c: Creating topic. name: cache:12/channel:212970, detail: [Aug 18 10:33:42] DEBUG[12890] stasis.c: Topic 'cache:12/channel:212970': 0x7f0c2c012960 created [Aug 18 10:33:42] DEBUG[12890] channel.c: Channel 0x7f0c2c010ca0 'SIP/zvonobot-00000005' allocated [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12890] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12890] res_stasis.c: calls_0: Subscribing to 212970 [Aug 18 10:33:42] DEBUG[12890] stasis/app.c: Channel '212970' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Outgoing Call for 79821117070 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12890] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12891] chan_sip.c: Audio is at 12972 [Aug 18 10:33:42] VERBOSE[12891] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] DEBUG[12890] http.c: HTTP closing session. Top level [Aug 18 10:33:42] VERBOSE[12891] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12891] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Initializing initreq for method INVITE - callid 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117070@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK463e7ded [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 3 [ 52]: From: ;tag=as0aff19ec [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 6 [ 60]: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12891] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117070@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK463e7ded Max-Forwards: 70 From: ;tag=as0aff19ec To: Contact: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 764687349 764687349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12972 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:42] DEBUG[12891] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12891] dial.c: Called zvonobot/79821117070 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK463e7ded;received=159.65.48.104 From: ;tag=as0aff19ec To: ;tag=as158ecc24 Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25361ae4" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK463e7ded;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0aff19ec [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as158ecc24 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25361ae4" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 (Checking To) --From tag as0aff19ec --To-tag as158ecc24 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117070@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK463e7ded Max-Forwards: 70 From: ;tag=as0aff19ec To: ;tag=as158ecc24 Contact: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 12972 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117070@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca Max-Forwards: 70 From: ;tag=as0aff19ec To: Contact: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117070@178.62.121.41", nonce="25361ae4", response="4cb534e3565d848e2dd6d97a8bd87229" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 764687349 764687350 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12972 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca;received=159.65.48.104 From: ;tag=as0aff19ec To: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0aff19ec [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 (Checking To) --From tag as0aff19ec --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12893] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12893] http.c: HTTP Request URI is /ari/channels/212969?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117071&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12893] http.c: match request [ari/channels/212969] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12893] http.c: match request [ari/channels/212969] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12893] http.c: match request [ari/channels/212969] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12893] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12893] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Finding handler for channels/212969 [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Finding handler for 212969 [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking channels create: Didn't match 212969 [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12893] res_ari.c: Checking channels externalMedia: Didn't match 212969 [Aug 18 10:33:42] DEBUG[12893] res_ari.c: No explicit handler found for 212969. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: Allocating new SIP dialog for 7ea07b7b2bb5f27c3054e9335bc862a5@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12893] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c34009e10' [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) RTP allocated port 11276 [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE creating session 0.0.0.0:11276 (11276) [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE create [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE add system candidates [Aug 18 10:33:42] DEBUG[12893] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12893] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE add candidate: 159.65.48.104:11276, 2130706431 [Aug 18 10:33:42] DEBUG[12893] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12893] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE add candidate: 10.131.0.10:11276, 2130706431 [Aug 18 10:33:42] DEBUG[12893] rtp_engine.c: RTP instance '0x7f0c34009e10' is setup and ready to go [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) ICE stopped [Aug 18 10:33:42] DEBUG[12893] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12893] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12893] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12893] res_rtp_asterisk.c: (0x7f0c34009e10) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12893] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12893] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: SIP call-id changed from '7ea07b7b2bb5f27c3054e9335bc862a5@127.0.1.1:5060' to '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12893] stasis.c: Creating topic. name: channel:212969, detail: [Aug 18 10:33:42] DEBUG[12893] stasis.c: Topic 'channel:212969': 0x7f0c34077e70 created [Aug 18 10:33:42] DEBUG[12893] stasis.c: Creating topic. name: cache:13/channel:212969, detail: [Aug 18 10:33:42] DEBUG[12893] stasis.c: Topic 'cache:13/channel:212969': 0x7f0c34013200 created [Aug 18 10:33:42] DEBUG[12893] channel.c: Channel 0x7f0c340114f0 'SIP/zvonobot-00000006' allocated [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12893] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12893] res_stasis.c: calls_0: Subscribing to 212969 [Aug 18 10:33:42] DEBUG[12893] stasis/app.c: Channel '212969' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Outgoing Call for 79821117071 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12893] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12894] chan_sip.c: Audio is at 11276 [Aug 18 10:33:42] VERBOSE[12894] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12894] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12894] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Initializing initreq for method INVITE - callid 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117071@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eb25c71 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 3 [ 52]: From: ;tag=as39a2ec19 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 6 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12893] http.c: HTTP closing session. Top level [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12894] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117071@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eb25c71 Max-Forwards: 70 From: ;tag=as39a2ec19 To: Contact: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168594803 1168594803 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11276 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:42] DEBUG[12894] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eb25c71;received=159.65.48.104 From: ;tag=as39a2ec19 To: ;tag=as4fb80756 Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1351bf1c" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eb25c71;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39a2ec19 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4fb80756 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1351bf1c" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 (Checking To) --From tag as39a2ec19 --To-tag as4fb80756 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117071@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eb25c71 Max-Forwards: 70 From: ;tag=as39a2ec19 To: ;tag=as4fb80756 Contact: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 11276 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117071@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323 Max-Forwards: 70 From: ;tag=as39a2ec19 To: Contact: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117071@178.62.121.41", nonce="1351bf1c", response="8bfbefb8cbf5d220e016cd274dcd57e7" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168594803 1168594804 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11276 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12894] dial.c: Called zvonobot/79821117071 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 From: ;tag=as39a2ec19 To: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39a2ec19 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 (Checking To) --From tag as39a2ec19 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12896] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12896] http.c: HTTP Request URI is /ari/channels/212971?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117069&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12896] http.c: match request [ari/channels/212971] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12896] http.c: match request [ari/channels/212971] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12896] http.c: match request [ari/channels/212971] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12896] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12896] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Finding handler for channels/212971 [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Finding handler for 212971 [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking channels create: Didn't match 212971 [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12896] res_ari.c: Checking channels externalMedia: Didn't match 212971 [Aug 18 10:33:42] DEBUG[12896] res_ari.c: No explicit handler found for 212971. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: Allocating new SIP dialog for 2d97d7b1305f9c774114c35c7fc3150c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12896] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c00b400' [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) RTP allocated port 11694 [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE creating session 0.0.0.0:11694 (11694) [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE create [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE add system candidates [Aug 18 10:33:42] DEBUG[12896] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12896] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE add candidate: 159.65.48.104:11694, 2130706431 [Aug 18 10:33:42] DEBUG[12896] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12896] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE add candidate: 10.131.0.10:11694, 2130706431 [Aug 18 10:33:42] DEBUG[12896] rtp_engine.c: RTP instance '0x7f0c3c00b400' is setup and ready to go [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE stopped [Aug 18 10:33:42] DEBUG[12896] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12896] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12896] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12896] res_rtp_asterisk.c: (0x7f0c3c00b400) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12896] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12896] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: SIP call-id changed from '2d97d7b1305f9c774114c35c7fc3150c@127.0.1.1:5060' to '3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12896] stasis.c: Creating topic. name: channel:212971, detail: [Aug 18 10:33:42] DEBUG[12896] stasis.c: Topic 'channel:212971': 0x7f0c3c011d40 created [Aug 18 10:33:42] DEBUG[12896] stasis.c: Creating topic. name: cache:14/channel:212971, detail: [Aug 18 10:33:42] DEBUG[12896] stasis.c: Topic 'cache:14/channel:212971': 0x7f0c3c011f00 created [Aug 18 10:33:42] DEBUG[12896] channel.c: Channel 0x7f0c3c0102e0 'SIP/zvonobot-00000007' allocated [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12896] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12896] res_stasis.c: calls_0: Subscribing to 212971 [Aug 18 10:33:42] DEBUG[12896] stasis/app.c: Channel '212971' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Outgoing Call for 79821117069 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12896] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12897] chan_sip.c: Audio is at 11694 [Aug 18 10:33:42] VERBOSE[12897] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12897] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12897] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Initializing initreq for method INVITE - callid 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117069@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK415aeaba [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 3 [ 52]: From: ;tag=as0f0e5c55 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 6 [ 60]: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12897] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117069@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK415aeaba Max-Forwards: 70 From: ;tag=as0f0e5c55 To: Contact: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 960493086 960493086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11694 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:42] DEBUG[12897] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12897] dial.c: Called zvonobot/79821117069 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] DEBUG[12896] http.c: HTTP closing session. Top level [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK415aeaba;received=159.65.48.104 From: ;tag=as0f0e5c55 To: ;tag=as602069b2 Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d96afb8" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK415aeaba;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0f0e5c55 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as602069b2 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d96afb8" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 (Checking To) --From tag as0f0e5c55 --To-tag as602069b2 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117069@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK415aeaba Max-Forwards: 70 From: ;tag=as0f0e5c55 To: ;tag=as602069b2 Contact: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 11694 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117069@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15 Max-Forwards: 70 From: ;tag=as0f0e5c55 To: Contact: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117069@178.62.121.41", nonce="2d96afb8", response="78a4e4cc74e63d22317c99e87558e656" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 960493086 960493087 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11694 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[12898] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12898] http.c: HTTP Request URI is /ari/channels/212972?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117068&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12898] http.c: match request [ari/channels/212972] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12898] http.c: match request [ari/channels/212972] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12898] http.c: match request [ari/channels/212972] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12898] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12898] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Finding handler for channels/212972 [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15;received=159.65.48.104 From: ;tag=as0f0e5c55 To: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Finding handler for 212972 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking channels create: Didn't match 212972 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0f0e5c55 [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[12898] res_ari.c: Checking channels externalMedia: Didn't match 212972 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12898] res_ari.c: No explicit handler found for 212972. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 (Checking To) --From tag as0f0e5c55 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: Allocating new SIP dialog for 54cd452e7ef82db6655800c335f99450@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12898] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c40006350' [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) RTP allocated port 17260 [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE creating session 0.0.0.0:17260 (17260) [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE create [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE add system candidates [Aug 18 10:33:42] DEBUG[12898] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12898] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE add candidate: 159.65.48.104:17260, 2130706431 [Aug 18 10:33:42] DEBUG[12898] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12898] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE add candidate: 10.131.0.10:17260, 2130706431 [Aug 18 10:33:42] DEBUG[12898] rtp_engine.c: RTP instance '0x7f0c40006350' is setup and ready to go [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) ICE stopped [Aug 18 10:33:42] DEBUG[12898] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12898] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12898] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12898] res_rtp_asterisk.c: (0x7f0c40006350) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12898] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12898] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: SIP call-id changed from '54cd452e7ef82db6655800c335f99450@127.0.1.1:5060' to '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12898] stasis.c: Creating topic. name: channel:212972, detail: [Aug 18 10:33:42] DEBUG[12898] stasis.c: Topic 'channel:212972': 0x7f0c40076ff0 created [Aug 18 10:33:42] DEBUG[12898] stasis.c: Creating topic. name: cache:15/channel:212972, detail: [Aug 18 10:33:42] DEBUG[12898] stasis.c: Topic 'cache:15/channel:212972': 0x7f0c400123a0 created [Aug 18 10:33:42] DEBUG[12898] channel.c: Channel 0x7f0c40010a50 'SIP/zvonobot-00000008' allocated [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12898] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12898] res_stasis.c: calls_0: Subscribing to 212972 [Aug 18 10:33:42] DEBUG[12898] stasis/app.c: Channel '212972' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Outgoing Call for 79821117068 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12898] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12900] chan_sip.c: Audio is at 17260 [Aug 18 10:33:42] VERBOSE[12900] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12900] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12900] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Initializing initreq for method INVITE - callid 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117068@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d9271fe [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 3 [ 52]: From: ;tag=as181bb145 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 6 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12900] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117068@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d9271fe Max-Forwards: 70 From: ;tag=as181bb145 To: Contact: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1725191917 1725191917 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:42] DEBUG[12900] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12900] dial.c: Called zvonobot/79821117068 [Aug 18 10:33:42] DEBUG[12898] http.c: HTTP closing session. Top level [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d9271fe;received=159.65.48.104 From: ;tag=as181bb145 To: ;tag=as17a2842e Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ccf4cd6" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d9271fe;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as181bb145 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as17a2842e [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ccf4cd6" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 (Checking To) --From tag as181bb145 --To-tag as17a2842e [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117068@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d9271fe Max-Forwards: 70 From: ;tag=as181bb145 To: ;tag=as17a2842e Contact: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 17260 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117068@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15c006b9 Max-Forwards: 70 From: ;tag=as181bb145 To: Contact: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117068@178.62.121.41", nonce="6ccf4cd6", response="3336cee7599a4d3fcf64a4a56ebe813b" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1725191917 1725191918 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15c006b9;received=159.65.48.104 From: ;tag=as181bb145 To: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15c006b9;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as181bb145 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 (Checking To) --From tag as181bb145 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:42] DEBUG[12902] http.c: HTTP opening session. Top level [Aug 18 10:33:42] DEBUG[12902] http.c: HTTP Request URI is /ari/channels/212973?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117067&callerId=74950493843 [Aug 18 10:33:42] DEBUG[12902] http.c: match request [ari/channels/212973] with handler [httpstatus] len 10 [Aug 18 10:33:42] DEBUG[12902] http.c: match request [ari/channels/212973] with handler [phoneprov] len 9 [Aug 18 10:33:42] DEBUG[12902] http.c: match request [ari/channels/212973] with handler [ari] len 3 [Aug 18 10:33:42] DEBUG[12902] http.c: Match made with [ari] [Aug 18 10:33:42] DEBUG[12902] http.c: HTTP consuming request body [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Finding handler for channels/212973 [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Finding handler for channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Finding handler for 212973 [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking channels create: Didn't match 212973 [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:42] DEBUG[12902] res_ari.c: Checking channels externalMedia: Didn't match 212973 [Aug 18 10:33:42] DEBUG[12902] res_ari.c: No explicit handler found for 212973. Using wildcard channelId. [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: Allocating new SIP dialog for 6b6070ca23730f8257687d6412e59bc5@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:42] DEBUG[12902] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c70012180' [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) RTP allocated port 16044 [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE creating session 0.0.0.0:16044 (16044) [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE create [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE add system candidates [Aug 18 10:33:42] DEBUG[12902] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:42] DEBUG[12902] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE add candidate: 159.65.48.104:16044, 2130706431 [Aug 18 10:33:42] DEBUG[12902] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:42] DEBUG[12902] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE add candidate: 10.131.0.10:16044, 2130706431 [Aug 18 10:33:42] DEBUG[12902] rtp_engine.c: RTP instance '0x7f0c70012180' is setup and ready to go [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) ICE stopped [Aug 18 10:33:42] DEBUG[12902] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:42] DEBUG[12902] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:42] DEBUG[12902] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:42] DEBUG[12902] res_rtp_asterisk.c: (0x7f0c70012180) RTCP setup on RTP instance [Aug 18 10:33:42] VERBOSE[12902] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:42] DEBUG[12902] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: SIP call-id changed from '6b6070ca23730f8257687d6412e59bc5@127.0.1.1:5060' to '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' [Aug 18 10:33:42] DEBUG[12902] stasis.c: Creating topic. name: channel:212973, detail: [Aug 18 10:33:42] DEBUG[12902] stasis.c: Topic 'channel:212973': 0x7f0c70080140 created [Aug 18 10:33:42] DEBUG[12902] stasis.c: Creating topic. name: cache:16/channel:212973, detail: [Aug 18 10:33:42] DEBUG[12902] stasis.c: Topic 'cache:16/channel:212973': 0x7f0c7007f5c0 created [Aug 18 10:33:42] DEBUG[12902] channel.c: Channel 0x7f0c700195c0 'SIP/zvonobot-00000009' allocated [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:42] DEBUG[12902] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:42] DEBUG[12902] res_stasis.c: calls_0: Subscribing to 212973 [Aug 18 10:33:42] DEBUG[12902] stasis/app.c: Channel '212973' is 1 interested in calls_0 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Outgoing Call for 79821117067 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:42] DEBUG[12902] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[12903] chan_sip.c: Audio is at 16044 [Aug 18 10:33:42] VERBOSE[12903] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[12903] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[12903] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Initializing initreq for method INVITE - callid 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117067@178.62.121.41 SIP/2.0 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK444cb056 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 3 [ 52]: From: ;tag=as0453a0d2 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 6 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:42 GMT [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:42] VERBOSE[12903] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117067@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK444cb056 Max-Forwards: 70 From: ;tag=as0453a0d2 To: Contact: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 634427030 634427030 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:42] DEBUG[12902] http.c: HTTP closing session. Top level [Aug 18 10:33:42] DEBUG[12903] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[12903] dial.c: Called zvonobot/79821117067 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:42] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK444cb056;received=159.65.48.104 From: ;tag=as0453a0d2 To: ;tag=as51794c37 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6abcd5d1" Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK444cb056;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0453a0d2 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as51794c37 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6abcd5d1" [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking To) --From tag as0453a0d2 --To-tag as51794c37 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Stopping retransmission on '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117067@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK444cb056 Max-Forwards: 70 From: ;tag=as0453a0d2 To: ;tag=as51794c37 Contact: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Audio is at 16044 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117067@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fbb5225 Max-Forwards: 70 From: ;tag=as0453a0d2 To: Contact: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117067@178.62.121.41", nonce="6abcd5d1", response="ff436152f7c3bd7c589d6c1289ea21aa" Date: Wed, 18 Aug 2021 10:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 634427030 634427031 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fbb5225;received=159.65.48.104 From: ;tag=as0453a0d2 To: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fbb5225;received=159.65.48.104 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0453a0d2 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:42] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking To) --From tag as0453a0d2 --To-tag [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:42] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12931] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12931] http.c: HTTP Request URI is /ari/channels/212974?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117066&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12931] http.c: match request [ari/channels/212974] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12931] http.c: match request [ari/channels/212974] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12931] http.c: match request [ari/channels/212974] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12931] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12931] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Finding handler for channels/212974 [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Finding handler for 212974 [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking channels create: Didn't match 212974 [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12931] res_ari.c: Checking channels externalMedia: Didn't match 212974 [Aug 18 10:33:44] DEBUG[12931] res_ari.c: No explicit handler found for 212974. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: Allocating new SIP dialog for 30900eb3782bb85f3d6bf52e7904fccb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12931] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7800c760' [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) RTP allocated port 11690 [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE creating session 0.0.0.0:11690 (11690) [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE create [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE add system candidates [Aug 18 10:33:44] DEBUG[12931] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12931] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE add candidate: 159.65.48.104:11690, 2130706431 [Aug 18 10:33:44] DEBUG[12931] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12931] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE add candidate: 10.131.0.10:11690, 2130706431 [Aug 18 10:33:44] DEBUG[12931] rtp_engine.c: RTP instance '0x7f0c7800c760' is setup and ready to go [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) ICE stopped [Aug 18 10:33:44] DEBUG[12931] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12931] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12931] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12931] res_rtp_asterisk.c: (0x7f0c7800c760) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12931] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12931] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: SIP call-id changed from '30900eb3782bb85f3d6bf52e7904fccb@127.0.1.1:5060' to '51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12931] stasis.c: Creating topic. name: channel:212974, detail: [Aug 18 10:33:44] DEBUG[12931] stasis.c: Topic 'channel:212974': 0x7f0c78013310 created [Aug 18 10:33:44] DEBUG[12931] stasis.c: Creating topic. name: cache:17/channel:212974, detail: [Aug 18 10:33:44] DEBUG[12931] stasis.c: Topic 'cache:17/channel:212974': 0x7f0c78013510 created [Aug 18 10:33:44] DEBUG[12931] channel.c: Channel 0x7f0c78011850 'SIP/zvonobot-0000000a' allocated [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12931] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12931] res_stasis.c: calls_0: Subscribing to 212974 [Aug 18 10:33:44] DEBUG[12931] stasis/app.c: Channel '212974' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Outgoing Call for 79821117066 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12931] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12933] chan_sip.c: Audio is at 11690 [Aug 18 10:33:44] VERBOSE[12933] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12933] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12933] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Initializing initreq for method INVITE - callid 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117066@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c92f62d [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 3 [ 52]: From: ;tag=as410f495a [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 6 [ 60]: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12933] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117066@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c92f62d Max-Forwards: 70 From: ;tag=as410f495a To: Contact: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1306294872 1306294872 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11690 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:44] DEBUG[12933] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12933] dial.c: Called zvonobot/79821117066 [Aug 18 10:33:44] DEBUG[12931] http.c: HTTP closing session. Top level [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c92f62d;received=159.65.48.104 From: ;tag=as410f495a To: ;tag=as7a6c42cc Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="791ffe59" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c92f62d;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as410f495a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7a6c42cc [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="791ffe59" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 (Checking To) --From tag as410f495a --To-tag as7a6c42cc [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117066@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c92f62d Max-Forwards: 70 From: ;tag=as410f495a To: ;tag=as7a6c42cc Contact: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 11690 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117066@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6b879d2d Max-Forwards: 70 From: ;tag=as410f495a To: Contact: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117066@178.62.121.41", nonce="791ffe59", response="28c5ea00b809538e256917cb60bde40b" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1306294872 1306294873 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11690 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6b879d2d;received=159.65.48.104 From: ;tag=as410f495a To: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6b879d2d;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as410f495a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 (Checking To) --From tag as410f495a --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12935] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12935] http.c: HTTP Request URI is /ari/channels/212975?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117065&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12935] http.c: match request [ari/channels/212975] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12935] http.c: match request [ari/channels/212975] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12935] http.c: match request [ari/channels/212975] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12935] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12935] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Finding handler for channels/212975 [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Finding handler for 212975 [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking channels create: Didn't match 212975 [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12935] res_ari.c: Checking channels externalMedia: Didn't match 212975 [Aug 18 10:33:44] DEBUG[12935] res_ari.c: No explicit handler found for 212975. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: Allocating new SIP dialog for 28b9bd8d08393a1b1bfb09913dee1ffa@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12935] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8000c760' [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) RTP allocated port 17072 [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE creating session 0.0.0.0:17072 (17072) [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE create [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE add system candidates [Aug 18 10:33:44] DEBUG[12935] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12935] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE add candidate: 159.65.48.104:17072, 2130706431 [Aug 18 10:33:44] DEBUG[12935] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12935] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE add candidate: 10.131.0.10:17072, 2130706431 [Aug 18 10:33:44] DEBUG[12935] rtp_engine.c: RTP instance '0x7f0c8000c760' is setup and ready to go [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) ICE stopped [Aug 18 10:33:44] DEBUG[12935] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12935] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12935] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12935] res_rtp_asterisk.c: (0x7f0c8000c760) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12935] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12935] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: SIP call-id changed from '28b9bd8d08393a1b1bfb09913dee1ffa@127.0.1.1:5060' to '36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12935] stasis.c: Creating topic. name: channel:212975, detail: [Aug 18 10:33:44] DEBUG[12935] stasis.c: Topic 'channel:212975': 0x7f0c80013310 created [Aug 18 10:33:44] DEBUG[12935] stasis.c: Creating topic. name: cache:18/channel:212975, detail: [Aug 18 10:33:44] DEBUG[12935] stasis.c: Topic 'cache:18/channel:212975': 0x7f0c80013510 created [Aug 18 10:33:44] DEBUG[12935] channel.c: Channel 0x7f0c80011850 'SIP/zvonobot-0000000b' allocated [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12935] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12935] res_stasis.c: calls_0: Subscribing to 212975 [Aug 18 10:33:44] DEBUG[12935] stasis/app.c: Channel '212975' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Outgoing Call for 79821117065 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12935] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12937] chan_sip.c: Audio is at 17072 [Aug 18 10:33:44] VERBOSE[12937] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12937] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12937] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Initializing initreq for method INVITE - callid 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117065@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0db6c3c5 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 3 [ 52]: From: ;tag=as42dc6c45 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 6 [ 60]: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] DEBUG[12935] http.c: HTTP closing session. Top level [Aug 18 10:33:44] VERBOSE[12937] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117065@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0db6c3c5 Max-Forwards: 70 From: ;tag=as42dc6c45 To: Contact: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1394389856 1394389856 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:44] DEBUG[12937] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12937] dial.c: Called zvonobot/79821117065 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0db6c3c5;received=159.65.48.104 From: ;tag=as42dc6c45 To: ;tag=as0239dcf1 Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7faa25de" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0db6c3c5;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc6c45 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0239dcf1 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7faa25de" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 (Checking To) --From tag as42dc6c45 --To-tag as0239dcf1 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117065@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0db6c3c5 Max-Forwards: 70 From: ;tag=as42dc6c45 To: ;tag=as0239dcf1 Contact: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 17072 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117065@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6 Max-Forwards: 70 From: ;tag=as42dc6c45 To: Contact: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117065@178.62.121.41", nonce="7faa25de", response="1f482e47ded2ba1e4a514818a757f09f" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1394389856 1394389857 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6;received=159.65.48.104 From: ;tag=as42dc6c45 To: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc6c45 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 (Checking To) --From tag as42dc6c45 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12940] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12940] http.c: HTTP Request URI is /ari/channels/212976?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117064&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12940] http.c: match request [ari/channels/212976] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12940] http.c: match request [ari/channels/212976] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12940] http.c: match request [ari/channels/212976] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12940] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12940] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Finding handler for channels/212976 [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Finding handler for 212976 [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking channels create: Didn't match 212976 [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12940] res_ari.c: Checking channels externalMedia: Didn't match 212976 [Aug 18 10:33:44] DEBUG[12940] res_ari.c: No explicit handler found for 212976. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: Allocating new SIP dialog for 7e083ceb6a6532fa39b15a7c4818a90b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12940] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8800fb50' [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) RTP allocated port 10330 [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE creating session 0.0.0.0:10330 (10330) [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE create [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE add system candidates [Aug 18 10:33:44] DEBUG[12940] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12940] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE add candidate: 159.65.48.104:10330, 2130706431 [Aug 18 10:33:44] DEBUG[12940] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12940] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE add candidate: 10.131.0.10:10330, 2130706431 [Aug 18 10:33:44] DEBUG[12940] rtp_engine.c: RTP instance '0x7f0c8800fb50' is setup and ready to go [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) ICE stopped [Aug 18 10:33:44] DEBUG[12940] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12940] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12940] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12940] res_rtp_asterisk.c: (0x7f0c8800fb50) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12940] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12940] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: SIP call-id changed from '7e083ceb6a6532fa39b15a7c4818a90b@127.0.1.1:5060' to '39299444695a01491d13a6704919adf8@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12940] stasis.c: Creating topic. name: channel:212976, detail: [Aug 18 10:33:44] DEBUG[12940] stasis.c: Topic 'channel:212976': 0x7f0c8807d960 created [Aug 18 10:33:44] DEBUG[12940] stasis.c: Creating topic. name: cache:19/channel:212976, detail: [Aug 18 10:33:44] DEBUG[12940] stasis.c: Topic 'cache:19/channel:212976': 0x7f0c8807db10 created [Aug 18 10:33:44] DEBUG[12940] channel.c: Channel 0x7f0c88017980 'SIP/zvonobot-0000000c' allocated [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12940] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12940] res_stasis.c: calls_0: Subscribing to 212976 [Aug 18 10:33:44] DEBUG[12940] stasis/app.c: Channel '212976' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Outgoing Call for 79821117064 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12940] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12942] chan_sip.c: Audio is at 10330 [Aug 18 10:33:44] VERBOSE[12942] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12942] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12942] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12940] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Initializing initreq for method INVITE - callid 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117064@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79cccb5d [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 3 [ 52]: From: ;tag=as42dc40dd [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 6 [ 60]: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12942] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117064@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79cccb5d Max-Forwards: 70 From: ;tag=as42dc40dd To: Contact: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1174639679 1174639679 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10330 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:44] DEBUG[12942] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12942] dial.c: Called zvonobot/79821117064 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79cccb5d;received=159.65.48.104 From: ;tag=as42dc40dd To: ;tag=as2b890500 Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="394b81a8" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79cccb5d;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc40dd [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2b890500 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="394b81a8" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 (Checking To) --From tag as42dc40dd --To-tag as2b890500 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '39299444695a01491d13a6704919adf8@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117064@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79cccb5d Max-Forwards: 70 From: ;tag=as42dc40dd To: ;tag=as2b890500 Contact: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 10330 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117064@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12 Max-Forwards: 70 From: ;tag=as42dc40dd To: Contact: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117064@178.62.121.41", nonce="394b81a8", response="4ac90f458a24cea6006b67d934f84ce5" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1174639679 1174639680 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10330 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12;received=159.65.48.104 From: ;tag=as42dc40dd To: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc40dd [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 (Checking To) --From tag as42dc40dd --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '39299444695a01491d13a6704919adf8@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12946] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12946] http.c: HTTP Request URI is /ari/channels/212979?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117061&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12946] http.c: match request [ari/channels/212979] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12946] http.c: match request [ari/channels/212979] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12946] http.c: match request [ari/channels/212979] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12946] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12946] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Finding handler for channels/212979 [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Finding handler for 212979 [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking channels create: Didn't match 212979 [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12946] res_ari.c: Checking channels externalMedia: Didn't match 212979 [Aug 18 10:33:44] DEBUG[12946] res_ari.c: No explicit handler found for 212979. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: Allocating new SIP dialog for 5d78c6c3607e368e59b4063a19dace0a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12946] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c90008240' [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) RTP allocated port 16840 [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE creating session 0.0.0.0:16840 (16840) [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE create [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE add system candidates [Aug 18 10:33:44] DEBUG[12946] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12946] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE add candidate: 159.65.48.104:16840, 2130706431 [Aug 18 10:33:44] DEBUG[12946] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12946] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE add candidate: 10.131.0.10:16840, 2130706431 [Aug 18 10:33:44] DEBUG[12946] rtp_engine.c: RTP instance '0x7f0c90008240' is setup and ready to go [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) ICE stopped [Aug 18 10:33:44] DEBUG[12946] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12946] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12946] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12946] res_rtp_asterisk.c: (0x7f0c90008240) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12946] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12946] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: SIP call-id changed from '5d78c6c3607e368e59b4063a19dace0a@127.0.1.1:5060' to '33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12946] stasis.c: Creating topic. name: channel:212979, detail: [Aug 18 10:33:44] DEBUG[12946] stasis.c: Topic 'channel:212979': 0x7f0c90011b80 created [Aug 18 10:33:44] DEBUG[12946] stasis.c: Creating topic. name: cache:20/channel:212979, detail: [Aug 18 10:33:44] DEBUG[12946] stasis.c: Topic 'cache:20/channel:212979': 0x7f0c90075c70 created [Aug 18 10:33:44] DEBUG[12946] channel.c: Channel 0x7f0c9000fab0 'SIP/zvonobot-0000000d' allocated [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12946] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12946] res_stasis.c: calls_0: Subscribing to 212979 [Aug 18 10:33:44] DEBUG[12946] stasis/app.c: Channel '212979' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Outgoing Call for 79821117061 [Aug 18 10:33:44] DEBUG[12946] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12948] chan_sip.c: Audio is at 16840 [Aug 18 10:33:44] VERBOSE[12948] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12948] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12948] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Initializing initreq for method INVITE - callid 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117061@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK019ef328 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 3 [ 52]: From: ;tag=as733c3052 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 6 [ 60]: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] DEBUG[12946] http.c: HTTP closing session. Top level [Aug 18 10:33:44] VERBOSE[12948] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117061@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK019ef328 Max-Forwards: 70 From: ;tag=as733c3052 To: Contact: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1886041958 1886041958 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:44] DEBUG[12948] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12948] dial.c: Called zvonobot/79821117061 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK019ef328;received=159.65.48.104 From: ;tag=as733c3052 To: ;tag=as0883a02b Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25c6bf49" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK019ef328;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as733c3052 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0883a02b [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25c6bf49" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 (Checking To) --From tag as733c3052 --To-tag as0883a02b [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117061@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK019ef328 Max-Forwards: 70 From: ;tag=as733c3052 To: ;tag=as0883a02b Contact: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 16840 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117061@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a05a098 Max-Forwards: 70 From: ;tag=as733c3052 To: Contact: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117061@178.62.121.41", nonce="25c6bf49", response="3237aa462185c1642f70c62405ebfd27" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1886041958 1886041959 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a05a098;received=159.65.48.104 From: ;tag=as733c3052 To: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[12952] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12952] http.c: HTTP Request URI is /ari/channels/212977?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117063&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12952] http.c: match request [ari/channels/212977] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12952] http.c: match request [ari/channels/212977] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[12952] http.c: match request [ari/channels/212977] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12952] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a05a098;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[12952] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as733c3052 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Finding handler for channels/212977 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Finding handler for 212977 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking channels create: Didn't match 212977 [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] DEBUG[12952] res_ari.c: Checking channels externalMedia: Didn't match 212977 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[12952] res_ari.c: No explicit handler found for 212977. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 (Checking To) --From tag as733c3052 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: Allocating new SIP dialog for 36256f260e31660a675d4eeb66e4b869@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12952] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c008a30' [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP allocated port 16132 [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE creating session 0.0.0.0:16132 (16132) [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE create [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE add system candidates [Aug 18 10:33:44] DEBUG[12952] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12952] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE add candidate: 159.65.48.104:16132, 2130706431 [Aug 18 10:33:44] DEBUG[12952] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12952] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE add candidate: 10.131.0.10:16132, 2130706431 [Aug 18 10:33:44] DEBUG[12952] rtp_engine.c: RTP instance '0x7f0c9c008a30' is setup and ready to go [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE stopped [Aug 18 10:33:44] DEBUG[12952] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12952] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12952] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12952] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12952] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12952] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: SIP call-id changed from '36256f260e31660a675d4eeb66e4b869@127.0.1.1:5060' to '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12952] stasis.c: Creating topic. name: channel:212977, detail: [Aug 18 10:33:44] DEBUG[12952] stasis.c: Topic 'channel:212977': 0x7f0c9c00f860 created [Aug 18 10:33:44] DEBUG[12952] stasis.c: Creating topic. name: cache:21/channel:212977, detail: [Aug 18 10:33:44] DEBUG[12952] stasis.c: Topic 'cache:21/channel:212977': 0x7f0c9c00f9b0 created [Aug 18 10:33:44] DEBUG[12952] channel.c: Channel 0x7f0c9c00dcf0 'SIP/zvonobot-0000000e' allocated [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12952] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12952] res_stasis.c: calls_0: Subscribing to 212977 [Aug 18 10:33:44] DEBUG[12952] stasis/app.c: Channel '212977' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Outgoing Call for 79821117063 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12952] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12956] chan_sip.c: Audio is at 16132 [Aug 18 10:33:44] VERBOSE[12956] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12956] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12956] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Initializing initreq for method INVITE - callid 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117063@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07305978 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 3 [ 52]: From: ;tag=as0b424b33 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 6 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12956] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117063@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07305978 Max-Forwards: 70 From: ;tag=as0b424b33 To: Contact: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861175660 1861175660 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:44] DEBUG[12956] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12956] dial.c: Called zvonobot/79821117063 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07305978;received=159.65.48.104 From: ;tag=as0b424b33 To: ;tag=as152341e6 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ea6bc94" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07305978;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0b424b33 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as152341e6 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ea6bc94" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 (Checking To) --From tag as0b424b33 --To-tag as152341e6 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117063@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07305978 Max-Forwards: 70 From: ;tag=as0b424b33 To: ;tag=as152341e6 Contact: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 16132 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117063@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02e5764a Max-Forwards: 70 From: ;tag=as0b424b33 To: Contact: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117063@178.62.121.41", nonce="1ea6bc94", response="fcb7a42d314b895bcf8f9992b25cd4b6" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861175660 1861175661 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[12952] http.c: HTTP closing session. Top level [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02e5764a;received=159.65.48.104 From: ;tag=as0b424b33 To: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02e5764a;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0b424b33 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 (Checking To) --From tag as0b424b33 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12958] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12958] http.c: HTTP Request URI is /ari/channels/212980?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117060&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12958] http.c: match request [ari/channels/212980] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12958] http.c: match request [ari/channels/212980] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12958] http.c: match request [ari/channels/212980] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12958] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12958] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Finding handler for channels/212980 [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Finding handler for 212980 [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking channels create: Didn't match 212980 [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12958] res_ari.c: Checking channels externalMedia: Didn't match 212980 [Aug 18 10:33:44] DEBUG[12958] res_ari.c: No explicit handler found for 212980. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: Allocating new SIP dialog for 396af08707aa23ac112492782cffce9e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12958] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca000a6f0' [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP allocated port 15402 [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE creating session 0.0.0.0:15402 (15402) [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE create [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE add system candidates [Aug 18 10:33:44] DEBUG[12958] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12958] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE add candidate: 159.65.48.104:15402, 2130706431 [Aug 18 10:33:44] DEBUG[12958] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12958] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE add candidate: 10.131.0.10:15402, 2130706431 [Aug 18 10:33:44] DEBUG[12958] rtp_engine.c: RTP instance '0x7f0ca000a6f0' is setup and ready to go [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE stopped [Aug 18 10:33:44] DEBUG[12958] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12958] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12958] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12958] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12958] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12958] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: SIP call-id changed from '396af08707aa23ac112492782cffce9e@127.0.1.1:5060' to '135d1bf70165f2237445fa857e02663b@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12958] stasis.c: Creating topic. name: channel:212980, detail: [Aug 18 10:33:44] DEBUG[12958] stasis.c: Topic 'channel:212980': 0x7f0ca0013680 created [Aug 18 10:33:44] DEBUG[12958] stasis.c: Creating topic. name: cache:22/channel:212980, detail: [Aug 18 10:33:44] DEBUG[12958] stasis.c: Topic 'cache:22/channel:212980': 0x7f0ca0013840 created [Aug 18 10:33:44] DEBUG[12958] channel.c: Channel 0x7f0ca0011c20 'SIP/zvonobot-0000000f' allocated [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12958] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12958] res_stasis.c: calls_0: Subscribing to 212980 [Aug 18 10:33:44] DEBUG[12958] stasis/app.c: Channel '212980' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Outgoing Call for 79821117060 [Aug 18 10:33:44] DEBUG[12958] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12958] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12959] chan_sip.c: Audio is at 15402 [Aug 18 10:33:44] VERBOSE[12959] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12959] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12959] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Initializing initreq for method INVITE - callid 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117060@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK294f696d [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 3 [ 52]: From: ;tag=as41697fdf [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 6 [ 60]: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12959] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117060@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK294f696d Max-Forwards: 70 From: ;tag=as41697fdf To: Contact: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904660760 904660760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15402 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:44] DEBUG[12959] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12959] dial.c: Called zvonobot/79821117060 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK294f696d;received=159.65.48.104 From: ;tag=as41697fdf To: ;tag=as24312329 Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c11d74" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK294f696d;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as41697fdf [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as24312329 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c11d74" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 (Checking To) --From tag as41697fdf --To-tag as24312329 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '135d1bf70165f2237445fa857e02663b@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117060@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK294f696d Max-Forwards: 70 From: ;tag=as41697fdf To: ;tag=as24312329 Contact: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 15402 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117060@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ffed886 Max-Forwards: 70 From: ;tag=as41697fdf To: Contact: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117060@178.62.121.41", nonce="23c11d74", response="4803d6e8f931ac18f631970a90ceb522" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904660760 904660761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15402 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ffed886;received=159.65.48.104 From: ;tag=as41697fdf To: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ffed886;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as41697fdf [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 (Checking To) --From tag as41697fdf --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '135d1bf70165f2237445fa857e02663b@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12961] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12961] http.c: HTTP Request URI is /ari/channels/212981?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117059&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12961] http.c: match request [ari/channels/212981] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12961] http.c: match request [ari/channels/212981] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12961] http.c: match request [ari/channels/212981] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12961] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12961] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Finding handler for channels/212981 [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Finding handler for 212981 [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking channels create: Didn't match 212981 [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12961] res_ari.c: Checking channels externalMedia: Didn't match 212981 [Aug 18 10:33:44] DEBUG[12961] res_ari.c: No explicit handler found for 212981. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: Allocating new SIP dialog for 1655326350f7c99732558b101da1dcd8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12961] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0015bc0' [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP allocated port 12102 [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE creating session 0.0.0.0:12102 (12102) [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE create [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE add system candidates [Aug 18 10:33:44] DEBUG[12961] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12961] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE add candidate: 159.65.48.104:12102, 2130706431 [Aug 18 10:33:44] DEBUG[12961] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12961] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE add candidate: 10.131.0.10:12102, 2130706431 [Aug 18 10:33:44] DEBUG[12961] rtp_engine.c: RTP instance '0x7f0cb0015bc0' is setup and ready to go [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE stopped [Aug 18 10:33:44] DEBUG[12961] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12961] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12961] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12961] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12961] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12961] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: SIP call-id changed from '1655326350f7c99732558b101da1dcd8@127.0.1.1:5060' to '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12961] stasis.c: Creating topic. name: channel:212981, detail: [Aug 18 10:33:44] DEBUG[12961] stasis.c: Topic 'channel:212981': 0x7f0cb00299b0 created [Aug 18 10:33:44] DEBUG[12961] stasis.c: Creating topic. name: cache:23/channel:212981, detail: [Aug 18 10:33:44] DEBUG[12961] stasis.c: Topic 'cache:23/channel:212981': 0x7f0cb00a4450 created [Aug 18 10:33:44] DEBUG[12961] channel.c: Channel 0x7f0cb003b730 'SIP/zvonobot-00000010' allocated [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12961] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12961] res_stasis.c: calls_0: Subscribing to 212981 [Aug 18 10:33:44] DEBUG[12961] stasis/app.c: Channel '212981' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Outgoing Call for 79821117059 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12961] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12961] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12962] chan_sip.c: Audio is at 12102 [Aug 18 10:33:44] VERBOSE[12962] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12962] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12962] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Initializing initreq for method INVITE - callid 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117059@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f1930e [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 3 [ 52]: From: ;tag=as4d3d785f [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 6 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12962] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117059@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f1930e Max-Forwards: 70 From: ;tag=as4d3d785f To: Contact: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 745273635 745273635 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12102 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:44] DEBUG[12962] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12962] dial.c: Called zvonobot/79821117059 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f1930e;received=159.65.48.104 From: ;tag=as4d3d785f To: ;tag=as1cead44a Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29c76a44" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f1930e;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d3d785f [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1cead44a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29c76a44" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 (Checking To) --From tag as4d3d785f --To-tag as1cead44a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117059@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f1930e Max-Forwards: 70 From: ;tag=as4d3d785f To: ;tag=as1cead44a Contact: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 12102 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117059@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b7e147e Max-Forwards: 70 From: ;tag=as4d3d785f To: Contact: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117059@178.62.121.41", nonce="29c76a44", response="6b3cfaf5a5c8972fde6ce1518382408d" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 745273635 745273636 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12102 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b7e147e;received=159.65.48.104 From: ;tag=as4d3d785f To: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b7e147e;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d3d785f [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 (Checking To) --From tag as4d3d785f --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12964] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12964] http.c: HTTP Request URI is /ari/channels/212978?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117062&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12964] http.c: match request [ari/channels/212978] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12964] http.c: match request [ari/channels/212978] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12964] http.c: match request [ari/channels/212978] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12964] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12964] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Finding handler for channels/212978 [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Finding handler for 212978 [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking channels create: Didn't match 212978 [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12964] res_ari.c: Checking channels externalMedia: Didn't match 212978 [Aug 18 10:33:44] DEBUG[12964] res_ari.c: No explicit handler found for 212978. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: Allocating new SIP dialog for 20f94b9761b3475e4189197a75eef775@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12964] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb401aa90' [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) RTP allocated port 12180 [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE creating session 0.0.0.0:12180 (12180) [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE create [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE add system candidates [Aug 18 10:33:44] DEBUG[12964] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12964] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE add candidate: 159.65.48.104:12180, 2130706431 [Aug 18 10:33:44] DEBUG[12964] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12964] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE add candidate: 10.131.0.10:12180, 2130706431 [Aug 18 10:33:44] DEBUG[12964] rtp_engine.c: RTP instance '0x7f0cb401aa90' is setup and ready to go [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE stopped [Aug 18 10:33:44] DEBUG[12964] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12964] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12964] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12964] res_rtp_asterisk.c: (0x7f0cb401aa90) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12964] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12964] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: SIP call-id changed from '20f94b9761b3475e4189197a75eef775@127.0.1.1:5060' to '5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12964] stasis.c: Creating topic. name: channel:212978, detail: [Aug 18 10:33:44] DEBUG[12964] stasis.c: Topic 'channel:212978': 0x7f0cb40211f0 created [Aug 18 10:33:44] DEBUG[12964] stasis.c: Creating topic. name: cache:24/channel:212978, detail: [Aug 18 10:33:44] DEBUG[12964] stasis.c: Topic 'cache:24/channel:212978': 0x7f0cb4023360 created [Aug 18 10:33:44] DEBUG[12964] channel.c: Channel 0x7f0cb401fdb0 'SIP/zvonobot-00000011' allocated [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12964] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12964] res_stasis.c: calls_0: Subscribing to 212978 [Aug 18 10:33:44] DEBUG[12964] stasis/app.c: Channel '212978' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Outgoing Call for 79821117062 [Aug 18 10:33:44] DEBUG[12964] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12965] chan_sip.c: Audio is at 12180 [Aug 18 10:33:44] VERBOSE[12965] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12965] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12965] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Initializing initreq for method INVITE - callid 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117062@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ec4ac50 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 3 [ 52]: From: ;tag=as123045f1 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 6 [ 60]: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12965] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117062@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ec4ac50 Max-Forwards: 70 From: ;tag=as123045f1 To: Contact: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 589627254 589627254 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12180 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:44] DEBUG[12965] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ec4ac50;received=159.65.48.104 From: ;tag=as123045f1 To: ;tag=as6d800516 Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5aae135a" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ec4ac50;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as123045f1 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6d800516 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5aae135a" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 (Checking To) --From tag as123045f1 --To-tag as6d800516 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117062@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ec4ac50 Max-Forwards: 70 From: ;tag=as123045f1 To: ;tag=as6d800516 Contact: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 12180 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117062@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f Max-Forwards: 70 From: ;tag=as123045f1 To: Contact: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117062@178.62.121.41", nonce="5aae135a", response="69f9b322ea22dcf91b057784e68f7878" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 589627254 589627255 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12180 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12965] dial.c: Called zvonobot/79821117062 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f;received=159.65.48.104 From: ;tag=as123045f1 To: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as123045f1 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 (Checking To) --From tag as123045f1 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12964] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12967] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12967] http.c: HTTP Request URI is /ari/channels/212983?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117057&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12967] http.c: match request [ari/channels/212983] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12967] http.c: match request [ari/channels/212983] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12967] http.c: match request [ari/channels/212983] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12967] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12967] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Finding handler for channels/212983 [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Finding handler for 212983 [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking channels create: Didn't match 212983 [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12967] res_ari.c: Checking channels externalMedia: Didn't match 212983 [Aug 18 10:33:44] DEBUG[12967] res_ari.c: No explicit handler found for 212983. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: Allocating new SIP dialog for 554c3a4372e393ab41fc87d27874e297@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12967] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1007bd40' [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP allocated port 11794 [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE creating session 0.0.0.0:11794 (11794) [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE create [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE add system candidates [Aug 18 10:33:44] DEBUG[12967] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12967] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE add candidate: 159.65.48.104:11794, 2130706431 [Aug 18 10:33:44] DEBUG[12967] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12967] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE add candidate: 10.131.0.10:11794, 2130706431 [Aug 18 10:33:44] DEBUG[12967] rtp_engine.c: RTP instance '0x7f0c1007bd40' is setup and ready to go [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE stopped [Aug 18 10:33:44] DEBUG[12967] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12967] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12967] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12967] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12967] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12967] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: SIP call-id changed from '554c3a4372e393ab41fc87d27874e297@127.0.1.1:5060' to '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12967] stasis.c: Creating topic. name: channel:212983, detail: [Aug 18 10:33:44] DEBUG[12967] stasis.c: Topic 'channel:212983': 0x7f0c100281f0 created [Aug 18 10:33:44] DEBUG[12967] stasis.c: Creating topic. name: cache:25/channel:212983, detail: [Aug 18 10:33:44] DEBUG[12967] stasis.c: Topic 'cache:25/channel:212983': 0x7f0c10026f10 created [Aug 18 10:33:44] DEBUG[12967] channel.c: Channel 0x7f0c10025b50 'SIP/zvonobot-00000012' allocated [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12967] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12967] res_stasis.c: calls_0: Subscribing to 212983 [Aug 18 10:33:44] DEBUG[12967] stasis/app.c: Channel '212983' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Outgoing Call for 79821117057 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12967] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12968] chan_sip.c: Audio is at 11794 [Aug 18 10:33:44] VERBOSE[12968] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12968] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12968] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Initializing initreq for method INVITE - callid 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117057@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK487e6b09 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 3 [ 52]: From: ;tag=as67678dc7 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 6 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12967] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12968] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117057@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK487e6b09 Max-Forwards: 70 From: ;tag=as67678dc7 To: Contact: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1174528635 1174528635 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11794 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:44] DEBUG[12968] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12968] dial.c: Called zvonobot/79821117057 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK487e6b09;received=159.65.48.104 From: ;tag=as67678dc7 To: ;tag=as7ef86daa Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08a4ff9b" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK487e6b09;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as67678dc7 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ef86daa [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08a4ff9b" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking To) --From tag as67678dc7 --To-tag as7ef86daa [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117057@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK487e6b09 Max-Forwards: 70 From: ;tag=as67678dc7 To: ;tag=as7ef86daa Contact: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 11794 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117057@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49f889c9 Max-Forwards: 70 From: ;tag=as67678dc7 To: Contact: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117057@178.62.121.41", nonce="08a4ff9b", response="f2c6a8d9649b6249cf3071e46d626730" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1174528635 1174528636 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11794 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49f889c9;received=159.65.48.104 From: ;tag=as67678dc7 To: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49f889c9;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as67678dc7 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking To) --From tag as67678dc7 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:44] DEBUG[12970] http.c: HTTP opening session. Top level [Aug 18 10:33:44] DEBUG[12970] http.c: HTTP Request URI is /ari/channels/212982?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117058&callerId=74950493843 [Aug 18 10:33:44] DEBUG[12970] http.c: match request [ari/channels/212982] with handler [httpstatus] len 10 [Aug 18 10:33:44] DEBUG[12970] http.c: match request [ari/channels/212982] with handler [phoneprov] len 9 [Aug 18 10:33:44] DEBUG[12970] http.c: match request [ari/channels/212982] with handler [ari] len 3 [Aug 18 10:33:44] DEBUG[12970] http.c: Match made with [ari] [Aug 18 10:33:44] DEBUG[12970] http.c: HTTP consuming request body [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Finding handler for channels/212982 [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Finding handler for channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Finding handler for 212982 [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking channels create: Didn't match 212982 [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:44] DEBUG[12970] res_ari.c: Checking channels externalMedia: Didn't match 212982 [Aug 18 10:33:44] DEBUG[12970] res_ari.c: No explicit handler found for 212982. Using wildcard channelId. [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: Allocating new SIP dialog for 01bbbfc3368958be65b6d8f61dcf3729@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:44] DEBUG[12970] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c00bc40' [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP allocated port 15966 [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE creating session 0.0.0.0:15966 (15966) [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE create [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE add system candidates [Aug 18 10:33:44] DEBUG[12970] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:44] DEBUG[12970] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE add candidate: 159.65.48.104:15966, 2130706431 [Aug 18 10:33:44] DEBUG[12970] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:44] DEBUG[12970] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE add candidate: 10.131.0.10:15966, 2130706431 [Aug 18 10:33:44] DEBUG[12970] rtp_engine.c: RTP instance '0x7f0c1c00bc40' is setup and ready to go [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE stopped [Aug 18 10:33:44] DEBUG[12970] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:44] DEBUG[12970] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:44] DEBUG[12970] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:44] DEBUG[12970] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP setup on RTP instance [Aug 18 10:33:44] VERBOSE[12970] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:44] DEBUG[12970] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: SIP call-id changed from '01bbbfc3368958be65b6d8f61dcf3729@127.0.1.1:5060' to '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' [Aug 18 10:33:44] DEBUG[12970] stasis.c: Creating topic. name: channel:212982, detail: [Aug 18 10:33:44] DEBUG[12970] stasis.c: Topic 'channel:212982': 0x7f0c1c028220 created [Aug 18 10:33:44] DEBUG[12970] stasis.c: Creating topic. name: cache:26/channel:212982, detail: [Aug 18 10:33:44] DEBUG[12970] stasis.c: Topic 'cache:26/channel:212982': 0x7f0c1c028740 created [Aug 18 10:33:44] DEBUG[12970] channel.c: Channel 0x7f0c1c026d20 'SIP/zvonobot-00000013' allocated [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:44] DEBUG[12970] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:44] DEBUG[12970] res_stasis.c: calls_0: Subscribing to 212982 [Aug 18 10:33:44] DEBUG[12970] stasis/app.c: Channel '212982' is 1 interested in calls_0 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Outgoing Call for 79821117058 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:44] DEBUG[12970] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[12971] chan_sip.c: Audio is at 15966 [Aug 18 10:33:44] VERBOSE[12971] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[12971] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[12971] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Initializing initreq for method INVITE - callid 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117058@178.62.121.41 SIP/2.0 [Aug 18 10:33:44] DEBUG[12970] http.c: HTTP closing session. Top level [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3843683a [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 3 [ 52]: From: ;tag=as574e1b12 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 6 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:44 GMT [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:44] VERBOSE[12971] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117058@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3843683a Max-Forwards: 70 From: ;tag=as574e1b12 To: Contact: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 991224472 991224472 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:44] DEBUG[12971] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[12971] dial.c: Called zvonobot/79821117058 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:44] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3843683a;received=159.65.48.104 From: ;tag=as574e1b12 To: ;tag=as7681cc1a Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26b444eb" Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3843683a;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as574e1b12 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7681cc1a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26b444eb" [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 (Checking To) --From tag as574e1b12 --To-tag as7681cc1a [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Stopping retransmission on '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117058@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3843683a Max-Forwards: 70 From: ;tag=as574e1b12 To: ;tag=as7681cc1a Contact: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Audio is at 15966 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117058@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab Max-Forwards: 70 From: ;tag=as574e1b12 To: Contact: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117058@178.62.121.41", nonce="26b444eb", response="8220e5cb51cbe41933f5346baad2218f" Date: Wed, 18 Aug 2021 10:33:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 991224472 991224473 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 From: ;tag=as574e1b12 To: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as574e1b12 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:44] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 (Checking To) --From tag as574e1b12 --To-tag [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:44] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:45] DEBUG[12981] http.c: HTTP opening session. Top level [Aug 18 10:33:45] DEBUG[12981] http.c: HTTP Request URI is /ari/channels/212984?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117056&callerId=74950493843 [Aug 18 10:33:45] DEBUG[12981] http.c: match request [ari/channels/212984] with handler [httpstatus] len 10 [Aug 18 10:33:45] DEBUG[12981] http.c: match request [ari/channels/212984] with handler [phoneprov] len 9 [Aug 18 10:33:45] DEBUG[12981] http.c: match request [ari/channels/212984] with handler [ari] len 3 [Aug 18 10:33:45] DEBUG[12981] http.c: Match made with [ari] [Aug 18 10:33:45] DEBUG[12981] http.c: HTTP consuming request body [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Finding handler for channels/212984 [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Finding handler for channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Finding handler for 212984 [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking channels create: Didn't match 212984 [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:45] DEBUG[12981] res_ari.c: Checking channels externalMedia: Didn't match 212984 [Aug 18 10:33:45] DEBUG[12981] res_ari.c: No explicit handler found for 212984. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: Allocating new SIP dialog for 170b305273d20fdb063adf0c740b2a85@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[12981] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c24007240' [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) RTP allocated port 10592 [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE creating session 0.0.0.0:10592 (10592) [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE create [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE add system candidates [Aug 18 10:33:46] DEBUG[12981] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[12981] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE add candidate: 159.65.48.104:10592, 2130706431 [Aug 18 10:33:46] DEBUG[12981] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[12981] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE add candidate: 10.131.0.10:10592, 2130706431 [Aug 18 10:33:46] DEBUG[12981] rtp_engine.c: RTP instance '0x7f0c24007240' is setup and ready to go [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) ICE stopped [Aug 18 10:33:46] DEBUG[12981] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[12981] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[12981] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[12981] res_rtp_asterisk.c: (0x7f0c24007240) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[12981] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[12981] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: SIP call-id changed from '170b305273d20fdb063adf0c740b2a85@127.0.1.1:5060' to '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[12981] stasis.c: Creating topic. name: channel:212984, detail: [Aug 18 10:33:46] DEBUG[12981] stasis.c: Topic 'channel:212984': 0x7f0c24025020 created [Aug 18 10:33:46] DEBUG[12981] stasis.c: Creating topic. name: cache:27/channel:212984, detail: [Aug 18 10:33:46] DEBUG[12981] stasis.c: Topic 'cache:27/channel:212984': 0x7f0c24023850 created [Aug 18 10:33:46] DEBUG[12981] channel.c: Channel 0x7f0c24021660 'SIP/zvonobot-00000014' allocated [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[12981] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[12981] res_stasis.c: calls_0: Subscribing to 212984 [Aug 18 10:33:46] DEBUG[12981] stasis/app.c: Channel '212984' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Outgoing Call for 79821117056 [Aug 18 10:33:46] DEBUG[12981] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[12982] chan_sip.c: Audio is at 10592 [Aug 18 10:33:46] VERBOSE[12982] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[12982] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[12982] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Initializing initreq for method INVITE - callid 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117056@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07b0806a [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 3 [ 52]: From: ;tag=as28b45d6b [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 6 [ 60]: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[12982] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117056@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07b0806a Max-Forwards: 70 From: ;tag=as28b45d6b To: Contact: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1435952064 1435952064 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10592 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:46] DEBUG[12982] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[12982] dial.c: Called zvonobot/79821117056 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07b0806a;received=159.65.48.104 From: ;tag=as28b45d6b To: ;tag=as4f4a6ace Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a66dcb3" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07b0806a;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28b45d6b [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4f4a6ace [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a66dcb3" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 (Checking To) --From tag as28b45d6b --To-tag as4f4a6ace [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117056@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07b0806a Max-Forwards: 70 From: ;tag=as28b45d6b To: ;tag=as4f4a6ace Contact: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 10592 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[12981] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117056@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67b43542 Max-Forwards: 70 From: ;tag=as28b45d6b To: Contact: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117056@178.62.121.41", nonce="0a66dcb3", response="92483f72a8afa4bc271f93143ccdf003" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1435952064 1435952065 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10592 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67b43542;received=159.65.48.104 From: ;tag=as28b45d6b To: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67b43542;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28b45d6b [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 (Checking To) --From tag as28b45d6b --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[12985] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[12985] http.c: HTTP Request URI is /ari/channels/212985?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117055&callerId=74950493843 [Aug 18 10:33:46] DEBUG[12985] http.c: match request [ari/channels/212985] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[12985] http.c: match request [ari/channels/212985] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[12985] http.c: match request [ari/channels/212985] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[12985] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[12985] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Finding handler for channels/212985 [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Finding handler for 212985 [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking channels create: Didn't match 212985 [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[12985] res_ari.c: Checking channels externalMedia: Didn't match 212985 [Aug 18 10:33:46] DEBUG[12985] res_ari.c: No explicit handler found for 212985. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: Allocating new SIP dialog for 75221190277934804810edf56ce98629@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[12985] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c01b720' [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) RTP allocated port 19072 [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE creating session 0.0.0.0:19072 (19072) [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE create [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE add system candidates [Aug 18 10:33:46] DEBUG[12985] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[12985] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE add candidate: 159.65.48.104:19072, 2130706431 [Aug 18 10:33:46] DEBUG[12985] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[12985] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE add candidate: 10.131.0.10:19072, 2130706431 [Aug 18 10:33:46] DEBUG[12985] rtp_engine.c: RTP instance '0x7f0c2c01b720' is setup and ready to go [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) ICE stopped [Aug 18 10:33:46] DEBUG[12985] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[12985] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[12985] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[12985] res_rtp_asterisk.c: (0x7f0c2c01b720) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[12985] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[12985] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: SIP call-id changed from '75221190277934804810edf56ce98629@127.0.1.1:5060' to '4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[12985] stasis.c: Creating topic. name: channel:212985, detail: [Aug 18 10:33:46] DEBUG[12985] stasis.c: Topic 'channel:212985': 0x7f0c2c022360 created [Aug 18 10:33:46] DEBUG[12985] stasis.c: Creating topic. name: cache:28/channel:212985, detail: [Aug 18 10:33:46] DEBUG[12985] stasis.c: Topic 'cache:28/channel:212985': 0x7f0c2c0240c0 created [Aug 18 10:33:46] DEBUG[12985] channel.c: Channel 0x7f0c2c020600 'SIP/zvonobot-00000015' allocated [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[12985] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[12985] res_stasis.c: calls_0: Subscribing to 212985 [Aug 18 10:33:46] DEBUG[12985] stasis/app.c: Channel '212985' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Outgoing Call for 79821117055 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[12985] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[12986] chan_sip.c: Audio is at 19072 [Aug 18 10:33:46] VERBOSE[12986] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[12986] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[12986] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Initializing initreq for method INVITE - callid 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117055@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK275b5c59 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 3 [ 52]: From: ;tag=as05f22381 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 6 [ 60]: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[12986] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117055@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK275b5c59 Max-Forwards: 70 From: ;tag=as05f22381 To: Contact: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2009425181 2009425181 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:46] DEBUG[12986] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[12986] dial.c: Called zvonobot/79821117055 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK275b5c59;received=159.65.48.104 From: ;tag=as05f22381 To: ;tag=as04c9b8ec Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21e87a1b" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK275b5c59;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as05f22381 [Aug 18 10:33:46] DEBUG[12985] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as04c9b8ec [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21e87a1b" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 (Checking To) --From tag as05f22381 --To-tag as04c9b8ec [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117055@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK275b5c59 Max-Forwards: 70 From: ;tag=as05f22381 To: ;tag=as04c9b8ec Contact: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 19072 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117055@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38c9360b Max-Forwards: 70 From: ;tag=as05f22381 To: Contact: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117055@178.62.121.41", nonce="21e87a1b", response="e24a9995bd9118a92fbcc0ed490af518" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2009425181 2009425182 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38c9360b;received=159.65.48.104 From: ;tag=as05f22381 To: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38c9360b;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as05f22381 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060 (Checking To) --From tag as05f22381 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4088907657f1fdef22eaee4f54c5cb75@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[12989] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[12989] http.c: HTTP Request URI is /ari/channels/212986?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117054&callerId=74950493843 [Aug 18 10:33:46] DEBUG[12989] http.c: match request [ari/channels/212986] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[12989] http.c: match request [ari/channels/212986] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[12989] http.c: match request [ari/channels/212986] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[12989] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[12989] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Finding handler for channels/212986 [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Finding handler for 212986 [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking channels create: Didn't match 212986 [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[12989] res_ari.c: Checking channels externalMedia: Didn't match 212986 [Aug 18 10:33:46] DEBUG[12989] res_ari.c: No explicit handler found for 212986. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: Allocating new SIP dialog for 40192216046f27b407dd4acb5063d48d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[12989] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3401c090' [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) RTP allocated port 17718 [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE creating session 0.0.0.0:17718 (17718) [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE create [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE add system candidates [Aug 18 10:33:46] DEBUG[12989] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[12989] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE add candidate: 159.65.48.104:17718, 2130706431 [Aug 18 10:33:46] DEBUG[12989] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[12989] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE add candidate: 10.131.0.10:17718, 2130706431 [Aug 18 10:33:46] DEBUG[12989] rtp_engine.c: RTP instance '0x7f0c3401c090' is setup and ready to go [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) ICE stopped [Aug 18 10:33:46] DEBUG[12989] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[12989] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[12989] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[12989] res_rtp_asterisk.c: (0x7f0c3401c090) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[12989] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[12989] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: SIP call-id changed from '40192216046f27b407dd4acb5063d48d@127.0.1.1:5060' to '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[12989] stasis.c: Creating topic. name: channel:212986, detail: [Aug 18 10:33:46] DEBUG[12989] stasis.c: Topic 'channel:212986': 0x7f0c34026ee0 created [Aug 18 10:33:46] DEBUG[12989] stasis.c: Creating topic. name: cache:29/channel:212986, detail: [Aug 18 10:33:46] DEBUG[12989] stasis.c: Topic 'cache:29/channel:212986': 0x7f0c34027950 created [Aug 18 10:33:46] DEBUG[12989] channel.c: Channel 0x7f0c34024da0 'SIP/zvonobot-00000016' allocated [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[12989] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[12989] res_stasis.c: calls_0: Subscribing to 212986 [Aug 18 10:33:46] DEBUG[12989] stasis/app.c: Channel '212986' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Outgoing Call for 79821117054 [Aug 18 10:33:46] DEBUG[12989] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[12989] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[12991] chan_sip.c: Audio is at 17718 [Aug 18 10:33:46] VERBOSE[12991] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[12991] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[12991] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Initializing initreq for method INVITE - callid 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117054@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK129932e7 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 3 [ 52]: From: ;tag=as14540915 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 6 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[12991] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117054@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK129932e7 Max-Forwards: 70 From: ;tag=as14540915 To: Contact: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 399603435 399603435 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17718 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:46] DEBUG[12991] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[12991] dial.c: Called zvonobot/79821117054 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK129932e7;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as4beaa126 Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="171898dd" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK129932e7;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4beaa126 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="171898dd" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as4beaa126 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117054@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK129932e7 Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as4beaa126 Contact: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 17718 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117054@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41960c20 Max-Forwards: 70 From: ;tag=as14540915 To: Contact: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41", nonce="171898dd", response="cde9f00043f485296925177993de6764" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 399603435 399603436 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17718 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41960c20;received=159.65.48.104 From: ;tag=as14540915 To: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41960c20;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[12994] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[12994] http.c: HTTP Request URI is /ari/channels/212987?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117053&callerId=74950493843 [Aug 18 10:33:46] DEBUG[12994] http.c: match request [ari/channels/212987] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[12994] http.c: match request [ari/channels/212987] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[12994] http.c: match request [ari/channels/212987] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[12994] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[12994] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Finding handler for channels/212987 [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Finding handler for 212987 [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking channels create: Didn't match 212987 [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[12994] res_ari.c: Checking channels externalMedia: Didn't match 212987 [Aug 18 10:33:46] DEBUG[12994] res_ari.c: No explicit handler found for 212987. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: Allocating new SIP dialog for 597874bd1d68786d67ef7f9070d3167e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[12994] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c005640' [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) RTP allocated port 14408 [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE creating session 0.0.0.0:14408 (14408) [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE create [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE add system candidates [Aug 18 10:33:46] DEBUG[12994] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[12994] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE add candidate: 159.65.48.104:14408, 2130706431 [Aug 18 10:33:46] DEBUG[12994] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[12994] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE add candidate: 10.131.0.10:14408, 2130706431 [Aug 18 10:33:46] DEBUG[12994] rtp_engine.c: RTP instance '0x7f0c3c005640' is setup and ready to go [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) ICE stopped [Aug 18 10:33:46] DEBUG[12994] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[12994] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[12994] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[12994] res_rtp_asterisk.c: (0x7f0c3c005640) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[12994] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[12994] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: SIP call-id changed from '597874bd1d68786d67ef7f9070d3167e@127.0.1.1:5060' to '7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[12994] stasis.c: Creating topic. name: channel:212987, detail: [Aug 18 10:33:46] DEBUG[12994] stasis.c: Topic 'channel:212987': 0x7f0c3c023ca0 created [Aug 18 10:33:46] DEBUG[12994] stasis.c: Creating topic. name: cache:30/channel:212987, detail: [Aug 18 10:33:46] DEBUG[12994] stasis.c: Topic 'cache:30/channel:212987': 0x7f0c3c01b9e0 created [Aug 18 10:33:46] DEBUG[12994] channel.c: Channel 0x7f0c3c020700 'SIP/zvonobot-00000017' allocated [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[12994] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[12994] res_stasis.c: calls_0: Subscribing to 212987 [Aug 18 10:33:46] DEBUG[12994] stasis/app.c: Channel '212987' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Outgoing Call for 79821117053 [Aug 18 10:33:46] DEBUG[12994] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13011] chan_sip.c: Audio is at 14408 [Aug 18 10:33:46] VERBOSE[13011] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13011] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13011] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Initializing initreq for method INVITE - callid 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117053@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096ae125 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 3 [ 52]: From: ;tag=as66bbd52b [Aug 18 10:33:46] DEBUG[12994] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 6 [ 60]: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13011] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117053@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096ae125 Max-Forwards: 70 From: ;tag=as66bbd52b To: Contact: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1121662359 1121662359 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14408 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:46] DEBUG[13011] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[13011] dial.c: Called zvonobot/79821117053 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096ae125;received=159.65.48.104 From: ;tag=as66bbd52b To: ;tag=as4ba80820 Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dac11a9" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096ae125;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as66bbd52b [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4ba80820 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dac11a9" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 (Checking To) --From tag as66bbd52b --To-tag as4ba80820 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117053@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096ae125 Max-Forwards: 70 From: ;tag=as66bbd52b To: ;tag=as4ba80820 Contact: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[13016] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13016] http.c: HTTP Request URI is /ari/channels/212990?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117050&callerId=74950493843 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13016] http.c: match request [ari/channels/212990] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] DEBUG[13016] http.c: match request [ari/channels/212990] with handler [phoneprov] len 9 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 14408 [Aug 18 10:33:46] DEBUG[13016] http.c: match request [ari/channels/212990] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13016] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13016] http.c: HTTP consuming request body [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Finding handler for channels/212990 [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Finding handler for channels [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Finding handler for 212990 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking channels create: Didn't match 212990 [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13016] res_ari.c: Checking channels externalMedia: Didn't match 212990 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13016] res_ari.c: No explicit handler found for 212990. Using wildcard channelId. [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117053@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2 Max-Forwards: 70 From: ;tag=as66bbd52b To: Contact: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117053@178.62.121.41", nonce="2dac11a9", response="96d267326bf4804dffdfe2a908bc2950" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1121662359 1121662360 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14408 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2;received=159.65.48.104 From: ;tag=as66bbd52b To: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as66bbd52b [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 (Checking To) --From tag as66bbd52b --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: Allocating new SIP dialog for 20cf741b44940b72086876f53a88144f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13016] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c40018bf0' [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) RTP allocated port 15650 [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE creating session 0.0.0.0:15650 (15650) [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE create [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE add system candidates [Aug 18 10:33:46] DEBUG[13016] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13016] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE add candidate: 159.65.48.104:15650, 2130706431 [Aug 18 10:33:46] DEBUG[13016] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13016] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE add candidate: 10.131.0.10:15650, 2130706431 [Aug 18 10:33:46] DEBUG[13016] rtp_engine.c: RTP instance '0x7f0c40018bf0' is setup and ready to go [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) ICE stopped [Aug 18 10:33:46] DEBUG[13016] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13016] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13016] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13016] res_rtp_asterisk.c: (0x7f0c40018bf0) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13016] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13016] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: SIP call-id changed from '20cf741b44940b72086876f53a88144f@127.0.1.1:5060' to '0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13016] stasis.c: Creating topic. name: channel:212990, detail: [Aug 18 10:33:46] DEBUG[13016] stasis.c: Topic 'channel:212990': 0x7f0c40023e90 created [Aug 18 10:33:46] DEBUG[13016] stasis.c: Creating topic. name: cache:31/channel:212990, detail: [Aug 18 10:33:46] DEBUG[13016] stasis.c: Topic 'cache:31/channel:212990': 0x7f0c40023f70 created [Aug 18 10:33:46] DEBUG[13016] channel.c: Channel 0x7f0c40020090 'SIP/zvonobot-00000018' allocated [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13016] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13016] res_stasis.c: calls_0: Subscribing to 212990 [Aug 18 10:33:46] DEBUG[13016] stasis/app.c: Channel '212990' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Outgoing Call for 79821117050 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13016] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13035] chan_sip.c: Audio is at 15650 [Aug 18 10:33:46] VERBOSE[13035] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13035] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13035] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Initializing initreq for method INVITE - callid 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117050@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c41e6bf [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 3 [ 52]: From: ;tag=as30f1f8f1 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 6 [ 60]: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13035] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117050@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c41e6bf Max-Forwards: 70 From: ;tag=as30f1f8f1 To: Contact: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895417031 1895417031 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15650 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:46] DEBUG[13035] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[13016] http.c: HTTP closing session. Top level [Aug 18 10:33:46] VERBOSE[13035] dial.c: Called zvonobot/79821117050 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c41e6bf;received=159.65.48.104 From: ;tag=as30f1f8f1 To: ;tag=as34d1dbb3 Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e30e836" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c41e6bf;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30f1f8f1 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34d1dbb3 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e30e836" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 (Checking To) --From tag as30f1f8f1 --To-tag as34d1dbb3 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117050@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c41e6bf Max-Forwards: 70 From: ;tag=as30f1f8f1 To: ;tag=as34d1dbb3 Contact: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 15650 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117050@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ca72f57 Max-Forwards: 70 From: ;tag=as30f1f8f1 To: Contact: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117050@178.62.121.41", nonce="1e30e836", response="033d393eeb835ae0789e682b6a39933a" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895417031 1895417032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15650 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[13037] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13037] http.c: HTTP Request URI is /ari/channels/212989?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117051&callerId=74950493843 [Aug 18 10:33:46] DEBUG[13037] http.c: match request [ari/channels/212989] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[13037] http.c: match request [ari/channels/212989] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[13037] http.c: match request [ari/channels/212989] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13037] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13037] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Finding handler for channels/212989 [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Finding handler for 212989 [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking channels create: Didn't match 212989 [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13037] res_ari.c: Checking channels externalMedia: Didn't match 212989 [Aug 18 10:33:46] DEBUG[13037] res_ari.c: No explicit handler found for 212989. Using wildcard channelId. [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ca72f57;received=159.65.48.104 From: ;tag=as30f1f8f1 To: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ca72f57;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30f1f8f1 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060 (Checking To) --From tag as30f1f8f1 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0089bb182110cb4c063ae02578941a5d@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: Allocating new SIP dialog for 183d1ba747cdc5f45536495913334500@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13037] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c70023de0' [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) RTP allocated port 12072 [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE creating session 0.0.0.0:12072 (12072) [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE create [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE add system candidates [Aug 18 10:33:46] DEBUG[13037] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13037] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE add candidate: 159.65.48.104:12072, 2130706431 [Aug 18 10:33:46] DEBUG[13037] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13037] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE add candidate: 10.131.0.10:12072, 2130706431 [Aug 18 10:33:46] DEBUG[13037] rtp_engine.c: RTP instance '0x7f0c70023de0' is setup and ready to go [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) ICE stopped [Aug 18 10:33:46] DEBUG[13037] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13037] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13037] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13037] res_rtp_asterisk.c: (0x7f0c70023de0) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13037] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13037] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: SIP call-id changed from '183d1ba747cdc5f45536495913334500@127.0.1.1:5060' to '72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13037] stasis.c: Creating topic. name: channel:212989, detail: [Aug 18 10:33:46] DEBUG[13037] stasis.c: Topic 'channel:212989': 0x7f0c7002a9c0 created [Aug 18 10:33:46] DEBUG[13037] stasis.c: Creating topic. name: cache:32/channel:212989, detail: [Aug 18 10:33:46] DEBUG[13037] stasis.c: Topic 'cache:32/channel:212989': 0x7f0c7002ac10 created [Aug 18 10:33:46] DEBUG[13037] channel.c: Channel 0x7f0c70029070 'SIP/zvonobot-00000019' allocated [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13037] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13037] res_stasis.c: calls_0: Subscribing to 212989 [Aug 18 10:33:46] DEBUG[13037] stasis/app.c: Channel '212989' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Outgoing Call for 79821117051 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13037] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13038] chan_sip.c: Audio is at 12072 [Aug 18 10:33:46] VERBOSE[13038] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13038] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13038] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Initializing initreq for method INVITE - callid 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117051@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70b93344 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 3 [ 52]: From: ;tag=as4a6e12c9 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 6 [ 60]: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13038] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117051@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70b93344 Max-Forwards: 70 From: ;tag=as4a6e12c9 To: Contact: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1765267257 1765267257 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:46] DEBUG[13038] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70b93344;received=159.65.48.104 From: ;tag=as4a6e12c9 To: ;tag=as25266be7 Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="741acd8f" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70b93344;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4a6e12c9 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as25266be7 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="741acd8f" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[13037] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 (Checking To) --From tag as4a6e12c9 --To-tag as25266be7 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117051@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70b93344 Max-Forwards: 70 From: ;tag=as4a6e12c9 To: ;tag=as25266be7 Contact: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 12072 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117051@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41ecf69e Max-Forwards: 70 From: ;tag=as4a6e12c9 To: Contact: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117051@178.62.121.41", nonce="741acd8f", response="68e258a6e0243c32d113c765dfa92157" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1765267257 1765267258 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[13038] dial.c: Called zvonobot/79821117051 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41ecf69e;received=159.65.48.104 From: ;tag=as4a6e12c9 To: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41ecf69e;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4a6e12c9 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060 (Checking To) --From tag as4a6e12c9 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '72cf21f84c83c646436dd56466b1cacb@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13040] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13040] http.c: HTTP Request URI is /ari/channels/212988?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117052&callerId=74950493843 [Aug 18 10:33:46] DEBUG[13040] http.c: match request [ari/channels/212988] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[13040] http.c: match request [ari/channels/212988] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[13040] http.c: match request [ari/channels/212988] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13040] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13040] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Finding handler for channels/212988 [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Finding handler for 212988 [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking channels create: Didn't match 212988 [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13040] res_ari.c: Checking channels externalMedia: Didn't match 212988 [Aug 18 10:33:46] DEBUG[13040] res_ari.c: No explicit handler found for 212988. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: Allocating new SIP dialog for 5cf182cb62f554f85922d05f3c2fd40c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13040] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c780068b0' [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) RTP allocated port 17320 [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE creating session 0.0.0.0:17320 (17320) [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE create [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE add system candidates [Aug 18 10:33:46] DEBUG[13040] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13040] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE add candidate: 159.65.48.104:17320, 2130706431 [Aug 18 10:33:46] DEBUG[13040] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13040] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE add candidate: 10.131.0.10:17320, 2130706431 [Aug 18 10:33:46] DEBUG[13040] rtp_engine.c: RTP instance '0x7f0c780068b0' is setup and ready to go [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) ICE stopped [Aug 18 10:33:46] DEBUG[13040] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13040] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13040] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13040] res_rtp_asterisk.c: (0x7f0c780068b0) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13040] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13040] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: SIP call-id changed from '5cf182cb62f554f85922d05f3c2fd40c@127.0.1.1:5060' to '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13040] stasis.c: Creating topic. name: channel:212988, detail: [Aug 18 10:33:46] DEBUG[13040] stasis.c: Topic 'channel:212988': 0x7f0c78022ac0 created [Aug 18 10:33:46] DEBUG[13040] stasis.c: Creating topic. name: cache:33/channel:212988, detail: [Aug 18 10:33:46] DEBUG[13040] stasis.c: Topic 'cache:33/channel:212988': 0x7f0c780226d0 created [Aug 18 10:33:46] DEBUG[13040] channel.c: Channel 0x7f0c78021200 'SIP/zvonobot-0000001a' allocated [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13040] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13040] res_stasis.c: calls_0: Subscribing to 212988 [Aug 18 10:33:46] DEBUG[13040] stasis/app.c: Channel '212988' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Outgoing Call for 79821117052 [Aug 18 10:33:46] DEBUG[13040] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13040] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13041] chan_sip.c: Audio is at 17320 [Aug 18 10:33:46] VERBOSE[13041] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13041] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13041] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Initializing initreq for method INVITE - callid 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117052@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5f02d7f3 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 3 [ 52]: From: ;tag=as404b2233 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 6 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13041] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117052@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5f02d7f3 Max-Forwards: 70 From: ;tag=as404b2233 To: Contact: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1561770118 1561770118 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17320 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:46] DEBUG[13041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[13041] dial.c: Called zvonobot/79821117052 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5f02d7f3;received=159.65.48.104 From: ;tag=as404b2233 To: ;tag=as331f4bdb Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f591e92" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5f02d7f3;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as404b2233 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as331f4bdb [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f591e92" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 (Checking To) --From tag as404b2233 --To-tag as331f4bdb [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117052@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5f02d7f3 Max-Forwards: 70 From: ;tag=as404b2233 To: ;tag=as331f4bdb Contact: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 17320 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117052@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e Max-Forwards: 70 From: ;tag=as404b2233 To: Contact: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117052@178.62.121.41", nonce="3f591e92", response="5d79135fef18ad6f91010d428f74fb90" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1561770118 1561770119 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17320 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 From: ;tag=as404b2233 To: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as404b2233 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 (Checking To) --From tag as404b2233 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13043] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13043] http.c: HTTP Request URI is /ari/channels/212993?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117047&callerId=74950493843 [Aug 18 10:33:46] DEBUG[13043] http.c: match request [ari/channels/212993] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[13043] http.c: match request [ari/channels/212993] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[13043] http.c: match request [ari/channels/212993] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13043] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13043] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Finding handler for channels/212993 [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Finding handler for 212993 [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking channels create: Didn't match 212993 [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13043] res_ari.c: Checking channels externalMedia: Didn't match 212993 [Aug 18 10:33:46] DEBUG[13043] res_ari.c: No explicit handler found for 212993. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: Allocating new SIP dialog for 023b9f6c20605ca810ed49643a24beec@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13043] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8001c6f0' [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTP allocated port 18850 [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE creating session 0.0.0.0:18850 (18850) [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE create [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE add system candidates [Aug 18 10:33:46] DEBUG[13043] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13043] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE add candidate: 159.65.48.104:18850, 2130706431 [Aug 18 10:33:46] DEBUG[13043] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13043] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE add candidate: 10.131.0.10:18850, 2130706431 [Aug 18 10:33:46] DEBUG[13043] rtp_engine.c: RTP instance '0x7f0c8001c6f0' is setup and ready to go [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE stopped [Aug 18 10:33:46] DEBUG[13043] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13043] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13043] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13043] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13043] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13043] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: SIP call-id changed from '023b9f6c20605ca810ed49643a24beec@127.0.1.1:5060' to '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13043] stasis.c: Creating topic. name: channel:212993, detail: [Aug 18 10:33:46] DEBUG[13043] stasis.c: Topic 'channel:212993': 0x7f0c80024080 created [Aug 18 10:33:46] DEBUG[13043] stasis.c: Creating topic. name: cache:34/channel:212993, detail: [Aug 18 10:33:46] DEBUG[13043] stasis.c: Topic 'cache:34/channel:212993': 0x7f0c80024280 created [Aug 18 10:33:46] DEBUG[13043] channel.c: Channel 0x7f0c800219b0 'SIP/zvonobot-0000001b' allocated [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13043] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13043] res_stasis.c: calls_0: Subscribing to 212993 [Aug 18 10:33:46] DEBUG[13043] stasis/app.c: Channel '212993' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Outgoing Call for 79821117047 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13043] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13044] chan_sip.c: Audio is at 18850 [Aug 18 10:33:46] VERBOSE[13044] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13044] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13044] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Initializing initreq for method INVITE - callid 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117047@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK53abb589 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 3 [ 52]: From: ;tag=as396a139d [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 6 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13044] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117047@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK53abb589 Max-Forwards: 70 From: ;tag=as396a139d To: Contact: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1181163771 1181163771 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18850 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:46] DEBUG[13044] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[13043] http.c: HTTP closing session. Top level [Aug 18 10:33:46] VERBOSE[13044] dial.c: Called zvonobot/79821117047 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK53abb589;received=159.65.48.104 From: ;tag=as396a139d To: ;tag=as07e00c47 Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39509751" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK53abb589;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as396a139d [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as07e00c47 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39509751" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 (Checking To) --From tag as396a139d --To-tag as07e00c47 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117047@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK53abb589 Max-Forwards: 70 From: ;tag=as396a139d To: ;tag=as07e00c47 Contact: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 18850 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117047@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2f56144c Max-Forwards: 70 From: ;tag=as396a139d To: Contact: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117047@178.62.121.41", nonce="39509751", response="d05d72cb525a319e1e2be70b7e88a817" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1181163771 1181163772 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18850 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2f56144c;received=159.65.48.104 From: ;tag=as396a139d To: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2f56144c;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as396a139d [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 (Checking To) --From tag as396a139d --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13046] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13046] http.c: HTTP Request URI is /ari/channels/212991?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117049&callerId=74950493843 [Aug 18 10:33:46] DEBUG[13046] http.c: match request [ari/channels/212991] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[13046] http.c: match request [ari/channels/212991] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[13046] http.c: match request [ari/channels/212991] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13046] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13046] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Finding handler for channels/212991 [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Finding handler for 212991 [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking channels create: Didn't match 212991 [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13046] res_ari.c: Checking channels externalMedia: Didn't match 212991 [Aug 18 10:33:46] DEBUG[13046] res_ari.c: No explicit handler found for 212991. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: Allocating new SIP dialog for 32dff448282398c612c6b3a3793a4366@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13046] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c88003e20' [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) RTP allocated port 16274 [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE creating session 0.0.0.0:16274 (16274) [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE create [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE add system candidates [Aug 18 10:33:46] DEBUG[13046] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13046] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE add candidate: 159.65.48.104:16274, 2130706431 [Aug 18 10:33:46] DEBUG[13046] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13046] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE add candidate: 10.131.0.10:16274, 2130706431 [Aug 18 10:33:46] DEBUG[13046] rtp_engine.c: RTP instance '0x7f0c88003e20' is setup and ready to go [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) ICE stopped [Aug 18 10:33:46] DEBUG[13046] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13046] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13046] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13046] res_rtp_asterisk.c: (0x7f0c88003e20) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13046] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13046] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: SIP call-id changed from '32dff448282398c612c6b3a3793a4366@127.0.1.1:5060' to '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13046] stasis.c: Creating topic. name: channel:212991, detail: [Aug 18 10:33:46] DEBUG[13046] stasis.c: Topic 'channel:212991': 0x7f0c8802ad00 created [Aug 18 10:33:46] DEBUG[13046] stasis.c: Creating topic. name: cache:35/channel:212991, detail: [Aug 18 10:33:46] DEBUG[13046] stasis.c: Topic 'cache:35/channel:212991': 0x7f0c88029ab0 created [Aug 18 10:33:46] DEBUG[13046] channel.c: Channel 0x7f0c880272f0 'SIP/zvonobot-0000001c' allocated [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13046] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13046] res_stasis.c: calls_0: Subscribing to 212991 [Aug 18 10:33:46] DEBUG[13046] stasis/app.c: Channel '212991' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Outgoing Call for 79821117049 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13046] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13047] chan_sip.c: Audio is at 16274 [Aug 18 10:33:46] VERBOSE[13047] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13047] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[13047] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Initializing initreq for method INVITE - callid 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117049@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27353073 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 3 [ 52]: From: ;tag=as0261f463 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 6 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13046] http.c: HTTP closing session. Top level [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13047] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117049@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27353073 Max-Forwards: 70 From: ;tag=as0261f463 To: Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1087437442 1087437442 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16274 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:46] DEBUG[13047] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[13047] dial.c: Called zvonobot/79821117049 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27353073;received=159.65.48.104 From: ;tag=as0261f463 To: ;tag=as078c1bf5 Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259bd5c0" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27353073;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0261f463 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as078c1bf5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259bd5c0" [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking To) --From tag as0261f463 --To-tag as078c1bf5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117049@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27353073 Max-Forwards: 70 From: ;tag=as0261f463 To: ;tag=as078c1bf5 Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 16274 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117049@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936 Max-Forwards: 70 From: ;tag=as0261f463 To: Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117049@178.62.121.41", nonce="259bd5c0", response="d1bfa50121f7a683d658366d37edf891" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1087437442 1087437443 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16274 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 From: ;tag=as0261f463 To: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0261f463 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking To) --From tag as0261f463 --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:46] DEBUG[13049] http.c: HTTP opening session. Top level [Aug 18 10:33:46] DEBUG[13049] http.c: HTTP Request URI is /ari/channels/212992?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117048&callerId=74950493843 [Aug 18 10:33:46] DEBUG[13049] http.c: match request [ari/channels/212992] with handler [httpstatus] len 10 [Aug 18 10:33:46] DEBUG[13049] http.c: match request [ari/channels/212992] with handler [phoneprov] len 9 [Aug 18 10:33:46] DEBUG[13049] http.c: match request [ari/channels/212992] with handler [ari] len 3 [Aug 18 10:33:46] DEBUG[13049] http.c: Match made with [ari] [Aug 18 10:33:46] DEBUG[13049] http.c: HTTP consuming request body [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Finding handler for channels/212992 [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Finding handler for channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Finding handler for 212992 [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking channels create: Didn't match 212992 [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:46] DEBUG[13049] res_ari.c: Checking channels externalMedia: Didn't match 212992 [Aug 18 10:33:46] DEBUG[13049] res_ari.c: No explicit handler found for 212992. Using wildcard channelId. [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: Allocating new SIP dialog for 76a21fb0531735310429c5db7114643d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:46] DEBUG[13049] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c900183c0' [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) RTP allocated port 13132 [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE creating session 0.0.0.0:13132 (13132) [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE create [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE add system candidates [Aug 18 10:33:46] DEBUG[13049] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:46] DEBUG[13049] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE add candidate: 159.65.48.104:13132, 2130706431 [Aug 18 10:33:46] DEBUG[13049] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:46] DEBUG[13049] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE add candidate: 10.131.0.10:13132, 2130706431 [Aug 18 10:33:46] DEBUG[13049] rtp_engine.c: RTP instance '0x7f0c900183c0' is setup and ready to go [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) ICE stopped [Aug 18 10:33:46] DEBUG[13049] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:46] DEBUG[13049] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:46] DEBUG[13049] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:46] DEBUG[13049] res_rtp_asterisk.c: (0x7f0c900183c0) RTCP setup on RTP instance [Aug 18 10:33:46] VERBOSE[13049] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:46] DEBUG[13049] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: SIP call-id changed from '76a21fb0531735310429c5db7114643d@127.0.1.1:5060' to '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' [Aug 18 10:33:46] DEBUG[13049] stasis.c: Creating topic. name: channel:212992, detail: [Aug 18 10:33:46] DEBUG[13049] stasis.c: Topic 'channel:212992': 0x7f0c900225d0 created [Aug 18 10:33:46] DEBUG[13049] stasis.c: Creating topic. name: cache:36/channel:212992, detail: [Aug 18 10:33:46] DEBUG[13049] stasis.c: Topic 'cache:36/channel:212992': 0x7f0c90021460 created [Aug 18 10:33:46] DEBUG[13049] channel.c: Channel 0x7f0c9001fe80 'SIP/zvonobot-0000001d' allocated [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:46] DEBUG[13049] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:46] DEBUG[13049] res_stasis.c: calls_0: Subscribing to 212992 [Aug 18 10:33:46] DEBUG[13049] stasis/app.c: Channel '212992' is 1 interested in calls_0 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Outgoing Call for 79821117048 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:46] DEBUG[13049] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[13050] chan_sip.c: Audio is at 13132 [Aug 18 10:33:46] VERBOSE[13050] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[13050] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] DEBUG[13049] http.c: HTTP closing session. Top level [Aug 18 10:33:46] VERBOSE[13050] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Initializing initreq for method INVITE - callid 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117048@178.62.121.41 SIP/2.0 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eab714b [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 3 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 6 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:46 GMT [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:46] VERBOSE[13050] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117048@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eab714b Max-Forwards: 70 From: ;tag=as57df1d1c To: Contact: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1161963425 1161963425 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:46] DEBUG[13050] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[13050] dial.c: Called zvonobot/79821117048 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eab714b;received=159.65.48.104 From: ;tag=as57df1d1c To: ;tag=as2091575a Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ec44b36" Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eab714b;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:33:46] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2091575a [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ec44b36" [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag as2091575a [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Stopping retransmission on '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117048@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1eab714b Max-Forwards: 70 From: ;tag=as57df1d1c To: ;tag=as2091575a Contact: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Audio is at 13132 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117048@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1 Max-Forwards: 70 From: ;tag=as57df1d1c To: Contact: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117048@178.62.121.41", nonce="4ec44b36", response="379a286661fafc09f17690eb1194296b" Date: Wed, 18 Aug 2021 10:33:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1161963425 1161963426 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 From: ;tag=as57df1d1c To: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:46] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:46] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK234c8021;received=159.65.48.104 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1606498569 1606498569 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11670 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK234c8021;received=159.65.48.104 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39d9ed01 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2c8bd98d [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1606498569 1606498569 IN IP4 178.62.121.41 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11670 RTP/AVP 0 8 101 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 (Checking To) --From tag as39d9ed01 --To-tag as2c8bd98d [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Stopping retransmission on '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Got SDP version 1606498569 and unique parts [root 1606498569 IN IP4 178.62.121.41] [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1606498569 1606498569 IN IP4 178.62.121.41... OK. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:47] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:47] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:47] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:47] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE set role failed; no ice instance [Aug 18 10:33:47] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP setting address on RTP instance [Aug 18 10:33:47] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0cb0010680 -- Strict RTP learning after remote address set to: 178.62.121.41:11670 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:11670 [Aug 18 10:33:47] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb003fbf8) from 0x7f0c147e2330 to 0x7f0cb0002148 [Aug 18 10:33:47] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00445d8) from 0x7f0c147e2330 to 0x7f0cb0002148 [Aug 18 10:33:47] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0048e78) from 0x7f0c147e2330 to 0x7f0cb0002148 [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP ignoring duplicate property [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:47] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000000 setting read format path: alaw -> alaw [Aug 18 10:33:47] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000000 setting write format path: alaw -> alaw [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS - ast_rtp_activate rtp=0x7f0cb0010680 - setup and perform DTLS' [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0010680) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0010680) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:47] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:47] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:47] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Strict routing enforced for session 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:47] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:47] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK554ec9bd Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Contact: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Session timer started: 16 - 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 1768000ms [Aug 18 10:33:47] VERBOSE[12865] dial.c: SIP/zvonobot-00000000 answered [Aug 18 10:33:47] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:47] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:47] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:47] VERBOSE[12865] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000000 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:47] DEBUG[12865] stasis/app.c: Channel '212964' is 2 interested in calls_0 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:47] DEBUG[13054] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13054] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:47] DEBUG[13054] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13054] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13054] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13054] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13054] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13054] stasis.c: Creating topic. name: bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c, detail: [Aug 18 10:33:47] DEBUG[13054] stasis.c: Topic 'bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c': 0x7f0c9c0130d0 created [Aug 18 10:33:47] DEBUG[13054] stasis.c: Creating topic. name: cache:37/bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c, detail: [Aug 18 10:33:47] DEBUG[13054] stasis.c: Topic 'cache:37/bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c': 0x7f0c9c002110 created [Aug 18 10:33:47] DEBUG[13054] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as two channels are required [Aug 18 10:33:47] DEBUG[13054] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13054] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13054] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:47] DEBUG[13054] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13054] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology constructor [Aug 18 10:33:47] DEBUG[13054] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology start [Aug 18 10:33:47] DEBUG[13054] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:47] DEBUG[13054] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13055] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13055] http.c: HTTP Request URI is /ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/addChannel?channel=212964 [Aug 18 10:33:47] DEBUG[13055] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13055] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13055] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/addChannel] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13055] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Finding handler for bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/addChannel [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Finding handler for 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13055] res_ari.c: No explicit handler found for 7421ba4f-6229-4eeb-b806-91ebc84ff38c. Using wildcard bridgeId. [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Finding handler for addChannel [Aug 18 10:33:47] DEBUG[13055] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:47] DEBUG[13055] stasis/control.c: 212964: Sending channel add_to_bridge command [Aug 18 10:33:47] DEBUG[12865] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000000 [Aug 18 10:33:47] DEBUG[12865] stasis/control.c: 212964: Adding to bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:33:47] DEBUG[12865] stasis/app.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' is 1 interested in calls_0 [Aug 18 10:33:47] DEBUG[13056] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining [Aug 18 10:33:47] DEBUG[13056] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pushing 0x7f0cac00a6f0(SIP/zvonobot-00000000) [Aug 18 10:33:47] VERBOSE[13056] bridge_channel.c: Channel SIP/zvonobot-00000000 joined 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:33:47] DEBUG[13056] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as two channels are required [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c is already using the new technology. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining simple_bridge technology [Aug 18 10:33:47] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP changing ssrc from 438079290 to 38951627 due to a source change [Aug 18 10:33:47] DEBUG[12865] stasis/app.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' is 2 interested in calls_0 [Aug 18 10:33:47] DEBUG[13055] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:47] DEBUG[13055] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13057] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13057] http.c: HTTP Request URI is /ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/record?name=212964_jkLJYorfMptPQaMaimvfjLSeLsoeTwlj&format=wav [Aug 18 10:33:47] DEBUG[13057] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/record] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13057] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/record] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13057] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/record] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13057] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Finding handler for bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Finding handler for 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13057] res_ari.c: No explicit handler found for 7421ba4f-6229-4eeb-b806-91ebc84ff38c. Using wildcard bridgeId. [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Finding handler for record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:47] DEBUG[13057] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:47] DEBUG[13057] stasis.c: Creating topic. name: channel:1629282827.30, detail: [Aug 18 10:33:47] DEBUG[13057] stasis.c: Topic 'channel:1629282827.30': 0x7f0ca4008c20 created [Aug 18 10:33:47] DEBUG[13057] stasis.c: Creating topic. name: cache:38/channel:1629282827.30, detail: [Aug 18 10:33:47] DEBUG[13057] stasis.c: Topic 'cache:38/channel:1629282827.30': 0x7f0ca40089a0 created [Aug 18 10:33:47] DEBUG[13057] channel.c: Channel 0x7f0ca4006700 'Recorder/ARI-00000000;1' allocated [Aug 18 10:33:47] DEBUG[13057] stasis.c: Creating topic. name: channel:1629282827.31, detail: [Aug 18 10:33:47] DEBUG[13057] stasis.c: Topic 'channel:1629282827.31': 0x7f0ca400e050 created [Aug 18 10:33:47] DEBUG[13057] stasis.c: Creating topic. name: cache:39/channel:1629282827.31, detail: [Aug 18 10:33:47] DEBUG[13057] stasis.c: Topic 'cache:39/channel:1629282827.31': 0x7f0ca400ffc0 created [Aug 18 10:33:47] DEBUG[13057] channel.c: Channel 0x7f0ca400e230 'Recorder/ARI-00000000;2' allocated [Aug 18 10:33:47] DEBUG[13057] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:47] DEBUG[13058] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining [Aug 18 10:33:47] DEBUG[13058] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pushing 0x7f0ca400f4f0(Recorder/ARI-00000000;2) [Aug 18 10:33:47] DEBUG[13058] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:47] VERBOSE[13058] bridge_channel.c: Channel Recorder/ARI-00000000;2 joined 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:33:47] DEBUG[13058] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c'. Checking compatability for channels 'SIP/zvonobot-00000000' and 'Recorder/ARI-00000000;2' [Aug 18 10:33:47] DEBUG[13058] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as could not get details [Aug 18 10:33:47] DEBUG[13058] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13058] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13058] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13058] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13058] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c is already using the new technology. [Aug 18 10:33:47] DEBUG[13058] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining simple_bridge technology [Aug 18 10:33:47] DEBUG[13058] channel.c: Channel Recorder/ARI-00000000;2 setting read format path: slin -> slin [Aug 18 10:33:47] DEBUG[13058] channel.c: Channel SIP/zvonobot-00000000 setting write format path: slin -> alaw [Aug 18 10:33:47] DEBUG[13058] channel.c: Channel SIP/zvonobot-00000000 setting read format path: alaw -> slin [Aug 18 10:33:47] DEBUG[13058] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:33:47] DEBUG[13057] res_stasis_recording.c: 1629282827.30: Sending record(212964_jkLJYorfMptPQaMaimvfjLSeLsoeTwlj.wav) command [Aug 18 10:33:47] DEBUG[13059] app.c: play_and_record: , /var/spool/asterisk/recording/212964_jkLJYorfMptPQaMaimvfjLSeLsoeTwlj, 'wav' [Aug 18 10:33:47] DEBUG[13057] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:47] DEBUG[13059] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:47] VERBOSE[13059] app.c: x=0, open writing: /var/spool/asterisk/recording/212964_jkLJYorfMptPQaMaimvfjLSeLsoeTwlj format: wav, 0x7f0cac012110 [Aug 18 10:33:47] DEBUG[13057] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13060] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13060] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:47] DEBUG[13060] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15;received=159.65.48.104 From: ;tag=as0f0e5c55 To: ;tag=as75741b21 Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15;received=159.65.48.104 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0f0e5c55 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as75741b21 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 (Checking To) --From tag as0f0e5c55 --To-tag as75741b21 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:47] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:47] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117069@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21093d15 Max-Forwards: 70 From: ;tag=as0f0e5c55 To: ;tag=as75741b21 Contact: Call-ID: 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:47] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:47] VERBOSE[12897] dial.c: SIP/zvonobot-00000007 is busy [Aug 18 10:33:47] DEBUG[12897] channel.c: Channel 0x7f0c3c0102e0 'SIP/zvonobot-00000007' hanging up. Refs: 2 [Aug 18 10:33:47] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000007 - start 1629282822.225283 answer 0.000000 end 1629282827.238551 dur 5.013 bill 1629282827.238 dispo BUSY [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:47] DEBUG[12897] chan_sip.c: Hangup call SIP/zvonobot-00000007, SIP callid 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:47] DEBUG[12897] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:47] DEBUG[12897] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:47] DEBUG[12897] channel.c: Channel 0x7f0c3c0102e0 'SIP/zvonobot-00000007' destroying [Aug 18 10:33:47] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282827.32, detail: [Aug 18 10:33:47] DEBUG[20545] stasis.c: Topic 'channel:1629282827.32': 0x7f0c300282d0 created [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:33:47] DEBUG[20620] stasis/app.c: channel '212971': is 0 interested in calls_0 [Aug 18 10:33:47] DEBUG[20620] stasis/app.c: channel '212971' unsubscribed from calls_0 [Aug 18 10:33:47] DEBUG[20545] stasis.c: Creating topic. name: cache:40/channel:1629282827.32, detail: [Aug 18 10:33:47] DEBUG[20545] stasis.c: Topic 'cache:40/channel:1629282827.32': 0x7f0c3002f680 created [Aug 18 10:33:47] DEBUG[20545] stasis.c: Destroying topic. name: cache:40/channel:1629282827.32, detail: [Aug 18 10:33:47] DEBUG[20545] stasis.c: Topic 'cache:40/channel:1629282827.32': 0x7f0c3002f680 destroyed [Aug 18 10:33:47] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282827.32, detail: [Aug 18 10:33:47] DEBUG[20545] stasis.c: Topic 'channel:1629282827.32': 0x7f0c300282d0 destroyed [Aug 18 10:33:47] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000007', '', 'AppDial2', '(Outgoing Line)', 5, 0, 'BUSY', 3, '', '212971', '')] [Aug 18 10:33:47] DEBUG[20523] threadpool.c: Increasing threadpool stasis/pool's size by 1 [Aug 18 10:33:47] DEBUG[13062] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13062] http.c: HTTP Request URI is /ari/channels/212971 [Aug 18 10:33:47] DEBUG[13062] http.c: match request [ari/channels/212971] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13062] http.c: match request [ari/channels/212971] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13062] http.c: match request [ari/channels/212971] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13062] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[12897] stasis.c: Destroying topic. name: cache:14/channel:212971, detail: [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Finding handler for channels/212971 [Aug 18 10:33:47] DEBUG[12897] stasis.c: Topic 'cache:14/channel:212971': 0x7f0c3c011f00 destroyed [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Finding handler for channels [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:47] DEBUG[12897] stasis.c: Destroying topic. name: channel:212971, detail: [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:47] DEBUG[12897] stasis.c: Topic 'channel:212971': 0x7f0c3c011d40 destroyed [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Finding handler for 212971 [Aug 18 10:33:47] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:47] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:47] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking channels create: Didn't match 212971 [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13062] res_ari.c: Checking channels externalMedia: Didn't match 212971 [Aug 18 10:33:47] DEBUG[13062] res_ari.c: No explicit handler found for 212971. Using wildcard channelId. [Aug 18 10:33:47] DEBUG[13060] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13060] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13060] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13062] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:47] DEBUG[13062] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13060] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13060] stasis.c: Creating topic. name: bridge:87d87304-31e6-4326-b367-680423189269, detail: [Aug 18 10:33:47] DEBUG[13060] stasis.c: Topic 'bridge:87d87304-31e6-4326-b367-680423189269': 0x7f0cb4027c50 created [Aug 18 10:33:47] DEBUG[13060] stasis.c: Creating topic. name: cache:41/bridge:87d87304-31e6-4326-b367-680423189269, detail: [Aug 18 10:33:47] DEBUG[13060] stasis.c: Topic 'cache:41/bridge:87d87304-31e6-4326-b367-680423189269': 0x7f0cb4012030 created [Aug 18 10:33:47] DEBUG[13060] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269' can not use native RTP bridge as two channels are required [Aug 18 10:33:47] DEBUG[13060] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13060] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13060] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:47] DEBUG[13060] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13060] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: calling simple_bridge technology constructor [Aug 18 10:33:47] DEBUG[13060] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: calling simple_bridge technology start [Aug 18 10:33:47] DEBUG[13060] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:47] DEBUG[13060] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13063] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13063] http.c: HTTP Request URI is /ari/channels/212964/snoop?app=calls_0&spy=in [Aug 18 10:33:47] DEBUG[13063] http.c: match request [ari/channels/212964/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13063] http.c: match request [ari/channels/212964/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13063] http.c: match request [ari/channels/212964/snoop] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13063] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Finding handler for channels/212964/snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Finding handler for channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Finding handler for 212964 [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channels create: Didn't match 212964 [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channels externalMedia: Didn't match 212964 [Aug 18 10:33:47] DEBUG[13063] res_ari.c: No explicit handler found for 212964. Using wildcard channelId. [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Finding handler for snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:47] DEBUG[13063] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:47] DEBUG[13063] stasis.c: Creating topic. name: channel:1629282827.33, detail: [Aug 18 10:33:47] DEBUG[13063] stasis.c: Topic 'channel:1629282827.33': 0x7f0c0800e710 created [Aug 18 10:33:47] DEBUG[13063] stasis.c: Creating topic. name: cache:42/channel:1629282827.33, detail: [Aug 18 10:33:47] DEBUG[13063] stasis.c: Topic 'cache:42/channel:1629282827.33': 0x7f0c0800e8f0 created [Aug 18 10:33:47] DEBUG[13063] channel.c: Channel 0x7f0c08011460 'Snoop/212964-00000000' allocated [Aug 18 10:33:47] DEBUG[13056] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c'. Checking compatability for channels 'SIP/zvonobot-00000000' and 'Recorder/ARI-00000000;2' [Aug 18 10:33:47] DEBUG[13056] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as channel 'SIP/zvonobot-00000000' has features which prevent it [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13063] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:47] DEBUG[13064] stasis/app.c: Channel '1629282827.33' is 1 interested in calls_0 [Aug 18 10:33:47] DEBUG[13063] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 457 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 457 [Aug 18 10:33:47] DEBUG[13056] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13056] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c is already using the new technology. [Aug 18 10:33:47] DEBUG[13069] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13069] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212964&app=calls_0&format=slin16&external_host=127.0.0.1%3A50394 [Aug 18 10:33:47] DEBUG[13069] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13069] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13069] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13069] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13067] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Finding handler for channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13069] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:47] DEBUG[13069] netsock2.c: Splitting '127.0.0.1:50394' into... [Aug 18 10:33:47] DEBUG[13067] http.c: HTTP Request URI is /ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play?media=sound%3Asilence%2F2 [Aug 18 10:33:47] DEBUG[13069] netsock2.c: ...host '127.0.0.1' and port '50394'. [Aug 18 10:33:47] DEBUG[13067] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13067] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13067] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13067] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13069] netsock2.c: Splitting '127.0.0.1:50394' into... [Aug 18 10:33:47] DEBUG[13069] netsock2.c: ...host '127.0.0.1' and port '50394'. [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Finding handler for bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13069] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13069] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c18009d50' [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Finding handler for 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) RTP allocated port 14606 [Aug 18 10:33:47] DEBUG[13067] res_ari.c: No explicit handler found for 7421ba4f-6229-4eeb-b806-91ebc84ff38c. Using wildcard bridgeId. [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Finding handler for play [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) ICE creating session 127.0.0.1:14606 (14606) [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) ICE create [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:47] DEBUG[13067] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:47] DEBUG[13067] stasis.c: Creating topic. name: channel:1629282827.34, detail: [Aug 18 10:33:47] DEBUG[13067] stasis.c: Topic 'channel:1629282827.34': 0x7f0c24006840 created [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) ICE add system candidates [Aug 18 10:33:47] DEBUG[13067] stasis.c: Creating topic. name: cache:43/channel:1629282827.34, detail: [Aug 18 10:33:47] DEBUG[13069] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:47] DEBUG[13069] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:47] DEBUG[13067] stasis.c: Topic 'cache:43/channel:1629282827.34': 0x7f0c240089d0 created [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) ICE add candidate: 159.65.48.104:14606, 2130706431 [Aug 18 10:33:47] DEBUG[13069] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:47] DEBUG[13069] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:47] DEBUG[13069] res_rtp_asterisk.c: (0x7f0c18009d50) ICE add candidate: 10.131.0.10:14606, 2130706431 [Aug 18 10:33:47] DEBUG[13069] rtp_engine.c: RTP instance '0x7f0c18009d50' is setup and ready to go [Aug 18 10:33:47] DEBUG[13069] stasis.c: Creating topic. name: channel:robot_212964, detail: [Aug 18 10:33:47] DEBUG[13067] channel.c: Channel 0x7f0c2402b670 'Announcer/ARI-00000001;1' allocated [Aug 18 10:33:47] DEBUG[13069] stasis.c: Topic 'channel:robot_212964': 0x7f0c18079310 created [Aug 18 10:33:47] DEBUG[13067] stasis.c: Creating topic. name: channel:1629282827.36, detail: [Aug 18 10:33:47] DEBUG[13069] stasis.c: Creating topic. name: cache:44/channel:robot_212964, detail: [Aug 18 10:33:47] DEBUG[13067] stasis.c: Topic 'channel:1629282827.36': 0x7f0c2402cbf0 created [Aug 18 10:33:47] DEBUG[13069] stasis.c: Topic 'cache:44/channel:robot_212964': 0x7f0c1807c280 created [Aug 18 10:33:47] DEBUG[13067] stasis.c: Creating topic. name: cache:45/channel:1629282827.36, detail: [Aug 18 10:33:47] DEBUG[13067] stasis.c: Topic 'cache:45/channel:1629282827.36': 0x7f0c24033210 created [Aug 18 10:33:47] DEBUG[13069] channel.c: Channel 0x7f0c18079740 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50' allocated [Aug 18 10:33:47] DEBUG[13067] channel.c: Channel 0x7f0c24031840 'Announcer/ARI-00000001;2' allocated [Aug 18 10:33:47] DEBUG[13069] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:47] VERBOSE[13069] res_rtp_asterisk.c: 0x7f0c1800feb0 -- Strict RTP learning after remote address set to: 127.0.0.1:50394 [Aug 18 10:33:47] DEBUG[13067] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:47] DEBUG[13067] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000001;1' [Aug 18 10:33:47] DEBUG[13069] res_stasis.c: calls_0: Subscribing to robot_212964 [Aug 18 10:33:47] DEBUG[13069] stasis/app.c: Channel 'robot_212964' is 1 interested in calls_0 [Aug 18 10:33:47] DEBUG[13071] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c240386d0(Announcer/ARI-00000001;2) is joining [Aug 18 10:33:47] DEBUG[13071] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pushing 0x7f0c240386d0(Announcer/ARI-00000001;2) [Aug 18 10:33:47] DEBUG[13069] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:47] DEBUG[13071] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:47] VERBOSE[13071] bridge_channel.c: Channel Announcer/ARI-00000001;2 joined 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13071] bridge.c: Chose bridge technology softmix [Aug 18 10:33:47] VERBOSE[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: switching from simple_bridge technology to softmix [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology constructor [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0cac00a6f0(SIP/zvonobot-00000000) to dummy bridge temporarily [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0ca400f4f0(Recorder/ARI-00000000;2) to dummy bridge temporarily [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is leaving simple_bridge technology (dummy) [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology stop [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c240386d0(Announcer/ARI-00000001;2) is joining softmix technology [Aug 18 10:33:47] DEBUG[13069] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Announcer/ARI-00000001;2: [Aug 18 10:33:47] DEBUG[13071] channel.c: Channel Announcer/ARI-00000001;2 setting write format path: slin -> slin [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Announcer/ARI-00000001;2: [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Announcer/ARI-00000001;2: Not in SFU mode [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining softmix technology [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: SIP/zvonobot-00000000: [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:47] VERBOSE[13072] dial.c: Called 127.0.0.1:50394 [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: SIP/zvonobot-00000000: [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: SIP/zvonobot-00000000: Not in SFU mode [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining softmix technology [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Recorder/ARI-00000000;2: [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:47] VERBOSE[13072] dial.c: UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 answered [Aug 18 10:33:47] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50394 [Aug 18 10:33:47] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50394 - state 2 (In use) [Aug 18 10:33:47] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50394, detail: [Aug 18 10:33:47] VERBOSE[13072] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 [Aug 18 10:33:47] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50394': 0x7f0c840260e0 created [Aug 18 10:33:47] DEBUG[13072] stasis/app.c: Channel 'robot_212964' is 2 interested in calls_0 [Aug 18 10:33:47] DEBUG[13071] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:47] DEBUG[13071] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: [Aug 18 10:33:47] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50394' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Recorder/ARI-00000000;2: [Aug 18 10:33:47] DEBUG[13071] bridge_softmix.c: Recorder/ARI-00000000;2: Not in SFU mode [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology start [Aug 18 10:33:47] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology destructor [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:47] DEBUG[13074] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13074] http.c: HTTP Request URI is /ari/bridges/87d87304-31e6-4326-b367-680423189269/addChannel?channel=1629282827.33%2Crobot_212964 [Aug 18 10:33:47] DEBUG[13074] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13074] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13074] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269/addChannel] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13074] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Finding handler for bridges/87d87304-31e6-4326-b367-680423189269/addChannel [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Finding handler for bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Finding handler for 87d87304-31e6-4326-b367-680423189269 [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13074] res_ari.c: No explicit handler found for 87d87304-31e6-4326-b367-680423189269. Using wildcard bridgeId. [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Finding handler for addChannel [Aug 18 10:33:47] DEBUG[13074] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:47] DEBUG[13073] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: starting mixing thread [Aug 18 10:33:47] DEBUG[13067] res_stasis_playback.c: 1629282827.34: Sending play(sound:silence/2) command [Aug 18 10:33:47] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:33:47] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP ooh, format changed from none to alaw [Aug 18 10:33:47] DEBUG[13067] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:47] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP starting transmission [Aug 18 10:33:47] DEBUG[13074] stasis/control.c: 1629282827.33: Sending channel add_to_bridge command [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:47] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:47] DEBUG[13075] channel.c: Channel Announcer/ARI-00000001;1 setting write format path: gsm -> slin [Aug 18 10:33:47] DEBUG[13075] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:47] VERBOSE[13075] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:47] DEBUG[13067] http.c: HTTP closing session. Top level [Aug 18 10:33:47] VERBOSE[13056] res_rtp_asterisk.c: 0x7f0cb0010680 -- Strict RTP switching to RTP target address 178.62.121.41:11670 as source [Aug 18 10:33:47] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:33:47] DEBUG[13064] bridge_roles.c: Roles did not exist on channel Snoop/212964-00000000 [Aug 18 10:33:47] DEBUG[13064] stasis/control.c: 1629282827.33: Adding to bridge 87d87304-31e6-4326-b367-680423189269 [Aug 18 10:33:47] DEBUG[13064] stasis/app.c: Bridge '87d87304-31e6-4326-b367-680423189269' is 1 interested in calls_0 [Aug 18 10:33:47] DEBUG[13076] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c1c00dbc0(Snoop/212964-00000000) is joining [Aug 18 10:33:47] DEBUG[13076] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: pushing 0x7f0c1c00dbc0(Snoop/212964-00000000) [Aug 18 10:33:47] VERBOSE[13076] bridge_channel.c: Channel Snoop/212964-00000000 joined 'simple_bridge' stasis-bridge <87d87304-31e6-4326-b367-680423189269> [Aug 18 10:33:47] DEBUG[13076] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269' can not use native RTP bridge as two channels are required [Aug 18 10:33:47] DEBUG[13076] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13076] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13076] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13076] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13076] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269 is already using the new technology. [Aug 18 10:33:47] DEBUG[13076] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c1c00dbc0(Snoop/212964-00000000) is joining simple_bridge technology [Aug 18 10:33:47] DEBUG[13064] stasis/app.c: Bridge '87d87304-31e6-4326-b367-680423189269' is 2 interested in calls_0 [Aug 18 10:33:47] DEBUG[13074] stasis/control.c: robot_212964: Sending channel add_to_bridge command [Aug 18 10:33:47] DEBUG[13072] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 [Aug 18 10:33:47] DEBUG[13072] stasis/control.c: robot_212964: Adding to bridge 87d87304-31e6-4326-b367-680423189269 [Aug 18 10:33:47] DEBUG[13072] stasis/app.c: Bridge '87d87304-31e6-4326-b367-680423189269' is 3 interested in calls_0 [Aug 18 10:33:47] DEBUG[13077] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) is joining [Aug 18 10:33:47] DEBUG[13077] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: pushing 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) [Aug 18 10:33:47] VERBOSE[13077] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 joined 'simple_bridge' stasis-bridge <87d87304-31e6-4326-b367-680423189269> [Aug 18 10:33:47] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 - start 1629282827.269978 answer 1629282827.275788 end 1629282827.478857 dur 0.208 bill 0.203 dispo ANSWERED [Aug 18 10:33:47] DEBUG[13077] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269'. Checking compatability for channels 'Snoop/212964-00000000' and 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50' [Aug 18 10:33:47] DEBUG[13077] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269' can not use native RTP bridge as could not get details [Aug 18 10:33:47] DEBUG[13077] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:47] DEBUG[13077] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13077] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:47] DEBUG[13077] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:47] DEBUG[13077] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269 is already using the new technology. [Aug 18 10:33:47] DEBUG[13077] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) is joining simple_bridge technology [Aug 18 10:33:47] DEBUG[13077] channel.c: Channel UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 setting read format path: slin16 -> slin16 [Aug 18 10:33:47] DEBUG[13077] channel.c: Channel Snoop/212964-00000000 setting write format path: slin16 -> slin [Aug 18 10:33:47] DEBUG[13077] channel.c: Channel Snoop/212964-00000000 setting read format path: slin -> slin16 [Aug 18 10:33:47] DEBUG[13077] channel.c: Channel UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 setting write format path: slin16 -> slin16 [Aug 18 10:33:47] DEBUG[13072] stasis/app.c: Bridge '87d87304-31e6-4326-b367-680423189269' is 4 interested in calls_0 [Aug 18 10:33:47] DEBUG[13074] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:47] DEBUG[13074] http.c: HTTP closing session. Top level [Aug 18 10:33:47] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP ooh, format changed from none to slin16 [Aug 18 10:33:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:47] VERBOSE[13086] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:47] VERBOSE[13092] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:47] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:47] VERBOSE[13099] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:47] DEBUG[13103] http.c: HTTP opening session. Top level [Aug 18 10:33:47] DEBUG[13103] http.c: HTTP Request URI is /ari/channels/212994?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117046&callerId=74950493843 [Aug 18 10:33:47] DEBUG[13103] http.c: match request [ari/channels/212994] with handler [httpstatus] len 10 [Aug 18 10:33:47] DEBUG[13103] http.c: match request [ari/channels/212994] with handler [phoneprov] len 9 [Aug 18 10:33:47] DEBUG[13103] http.c: match request [ari/channels/212994] with handler [ari] len 3 [Aug 18 10:33:47] DEBUG[13103] http.c: Match made with [ari] [Aug 18 10:33:47] DEBUG[13103] http.c: HTTP consuming request body [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Finding handler for channels/212994 [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Finding handler for channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Finding handler for 212994 [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking channels create: Didn't match 212994 [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:47] DEBUG[13103] res_ari.c: Checking channels externalMedia: Didn't match 212994 [Aug 18 10:33:47] DEBUG[13103] res_ari.c: No explicit handler found for 212994. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: Allocating new SIP dialog for 2274d12a312809fa26741ba22a018447@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13103] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c00e8e0' [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) RTP allocated port 10722 [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE creating session 0.0.0.0:10722 (10722) [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE create [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE add system candidates [Aug 18 10:33:48] DEBUG[13103] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13103] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE add candidate: 159.65.48.104:10722, 2130706431 [Aug 18 10:33:48] DEBUG[13103] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13103] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE add candidate: 10.131.0.10:10722, 2130706431 [Aug 18 10:33:48] DEBUG[13103] rtp_engine.c: RTP instance '0x7f0c7c00e8e0' is setup and ready to go [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE stopped [Aug 18 10:33:48] DEBUG[13103] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13103] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13103] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13103] res_rtp_asterisk.c: (0x7f0c7c00e8e0) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13103] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13103] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: SIP call-id changed from '2274d12a312809fa26741ba22a018447@127.0.1.1:5060' to '33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13103] stasis.c: Creating topic. name: channel:212994, detail: [Aug 18 10:33:48] DEBUG[13103] stasis.c: Topic 'channel:212994': 0x7f0c7c07c040 created [Aug 18 10:33:48] DEBUG[13103] stasis.c: Creating topic. name: cache:46/channel:212994, detail: [Aug 18 10:33:48] DEBUG[13103] stasis.c: Topic 'cache:46/channel:212994': 0x7f0c7c0176e0 created [Aug 18 10:33:48] DEBUG[13103] channel.c: Channel 0x7f0c7c015960 'SIP/zvonobot-0000001e' allocated [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13103] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3d852a430704f3c8666a17646b0d1921@159.65.48.104:5060' Method: INVITE [Aug 18 10:33:48] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS stop [Aug 18 10:33:48] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:48] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) DTLS srtp - stopped timeout timer' [Aug 18 10:33:48] DEBUG[13103] res_stasis.c: calls_0: Subscribing to 212994 [Aug 18 10:33:48] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c00b400) ICE RTP transport deallocating [Aug 18 10:33:48] DEBUG[13103] stasis/app.c: Channel '212994' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3c00b400' [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Outgoing Call for 79821117046 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13103] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13105] chan_sip.c: Audio is at 10722 [Aug 18 10:33:48] VERBOSE[13105] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13105] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13105] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Initializing initreq for method INVITE - callid 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117046@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c8eeb21 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 3 [ 52]: From: ;tag=as16e0fe9d [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 6 [ 60]: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13103] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13105] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117046@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c8eeb21 Max-Forwards: 70 From: ;tag=as16e0fe9d To: Contact: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 717696246 717696246 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10722 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:48] DEBUG[13105] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13105] dial.c: Called zvonobot/79821117046 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c8eeb21;received=159.65.48.104 From: ;tag=as16e0fe9d To: ;tag=as06508b87 Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="194aa365" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c8eeb21;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as16e0fe9d [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as06508b87 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="194aa365" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 (Checking To) --From tag as16e0fe9d --To-tag as06508b87 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117046@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c8eeb21 Max-Forwards: 70 From: ;tag=as16e0fe9d To: ;tag=as06508b87 Contact: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 10722 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117046@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36 Max-Forwards: 70 From: ;tag=as16e0fe9d To: Contact: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117046@178.62.121.41", nonce="194aa365", response="2cdadf0bf0cc48794935a593ec4f9fe4" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 717696246 717696247 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10722 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36;received=159.65.48.104 From: ;tag=as16e0fe9d To: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as16e0fe9d [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 (Checking To) --From tag as16e0fe9d --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13108] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13108] http.c: HTTP Request URI is /ari/channels/212996?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117044&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13108] http.c: match request [ari/channels/212996] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13108] http.c: match request [ari/channels/212996] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13108] http.c: match request [ari/channels/212996] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13108] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13108] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Finding handler for channels/212996 [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Finding handler for 212996 [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking channels create: Didn't match 212996 [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13108] res_ari.c: Checking channels externalMedia: Didn't match 212996 [Aug 18 10:33:48] DEBUG[13108] res_ari.c: No explicit handler found for 212996. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: Allocating new SIP dialog for 66d312e45ea0cc601a8799d001a48f5a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13108] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8402e520' [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) RTP allocated port 15096 [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE creating session 0.0.0.0:15096 (15096) [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE create [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE add system candidates [Aug 18 10:33:48] DEBUG[13108] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13108] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE add candidate: 159.65.48.104:15096, 2130706431 [Aug 18 10:33:48] DEBUG[13108] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13108] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE add candidate: 10.131.0.10:15096, 2130706431 [Aug 18 10:33:48] DEBUG[13108] rtp_engine.c: RTP instance '0x7f0c8402e520' is setup and ready to go [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) ICE stopped [Aug 18 10:33:48] DEBUG[13108] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13108] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13108] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13108] res_rtp_asterisk.c: (0x7f0c8402e520) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13108] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13108] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: SIP call-id changed from '66d312e45ea0cc601a8799d001a48f5a@127.0.1.1:5060' to '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13108] stasis.c: Creating topic. name: channel:212996, detail: [Aug 18 10:33:48] DEBUG[13108] stasis.c: Topic 'channel:212996': 0x7f0c8409b630 created [Aug 18 10:33:48] DEBUG[13108] stasis.c: Creating topic. name: cache:47/channel:212996, detail: [Aug 18 10:33:48] DEBUG[13108] stasis.c: Topic 'cache:47/channel:212996': 0x7f0c8409b810 created [Aug 18 10:33:48] DEBUG[13108] channel.c: Channel 0x7f0c84035ac0 'SIP/zvonobot-0000001f' allocated [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13108] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13108] res_stasis.c: calls_0: Subscribing to 212996 [Aug 18 10:33:48] DEBUG[13108] stasis/app.c: Channel '212996' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Outgoing Call for 79821117044 [Aug 18 10:33:48] DEBUG[13108] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13109] chan_sip.c: Audio is at 15096 [Aug 18 10:33:48] VERBOSE[13109] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13109] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13109] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Initializing initreq for method INVITE - callid 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117044@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50c69ca8 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 3 [ 52]: From: ;tag=as31e40966 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 6 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13109] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117044@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50c69ca8 Max-Forwards: 70 From: ;tag=as31e40966 To: Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 383593663 383593663 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15096 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:48] DEBUG[13109] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[13108] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50c69ca8;received=159.65.48.104 From: ;tag=as31e40966 To: ;tag=as2a378a51 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34e58402" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50c69ca8;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as31e40966 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2a378a51 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34e58402" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking To) --From tag as31e40966 --To-tag as2a378a51 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117044@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50c69ca8 Max-Forwards: 70 From: ;tag=as31e40966 To: ;tag=as2a378a51 Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13109] dial.c: Called zvonobot/79821117044 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 15096 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117044@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885 Max-Forwards: 70 From: ;tag=as31e40966 To: Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117044@178.62.121.41", nonce="34e58402", response="0c845c46ef8d23b5df4e976bf6d77eb8" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 383593663 383593664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15096 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 From: ;tag=as31e40966 To: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as31e40966 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking To) --From tag as31e40966 --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #2 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13112] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13112] http.c: HTTP Request URI is /ari/channels/212995?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117045&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13112] http.c: match request [ari/channels/212995] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13112] http.c: match request [ari/channels/212995] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13112] http.c: match request [ari/channels/212995] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13112] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13112] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Finding handler for channels/212995 [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Finding handler for 212995 [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking channels create: Didn't match 212995 [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13112] res_ari.c: Checking channels externalMedia: Didn't match 212995 [Aug 18 10:33:48] DEBUG[13112] res_ari.c: No explicit handler found for 212995. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: Allocating new SIP dialog for 0a493338747c84c106531ce511563bd9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13112] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c00f7e0' [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP allocated port 17318 [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE creating session 0.0.0.0:17318 (17318) [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE create [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE add system candidates [Aug 18 10:33:48] DEBUG[13112] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13112] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE add candidate: 159.65.48.104:17318, 2130706431 [Aug 18 10:33:48] DEBUG[13112] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13112] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE add candidate: 10.131.0.10:17318, 2130706431 [Aug 18 10:33:48] DEBUG[13112] rtp_engine.c: RTP instance '0x7f0c8c00f7e0' is setup and ready to go [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE stopped [Aug 18 10:33:48] DEBUG[13112] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13112] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13112] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13112] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13112] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13112] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: SIP call-id changed from '0a493338747c84c106531ce511563bd9@127.0.1.1:5060' to '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13112] stasis.c: Creating topic. name: channel:212995, detail: [Aug 18 10:33:48] DEBUG[13112] stasis.c: Topic 'channel:212995': 0x7f0c8c0199f0 created [Aug 18 10:33:48] DEBUG[13112] stasis.c: Creating topic. name: cache:48/channel:212995, detail: [Aug 18 10:33:48] DEBUG[13112] stasis.c: Topic 'cache:48/channel:212995': 0x7f0c8c07da20 created [Aug 18 10:33:48] DEBUG[13112] channel.c: Channel 0x7f0c8c0178b0 'SIP/zvonobot-00000020' allocated [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13112] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13112] res_stasis.c: calls_0: Subscribing to 212995 [Aug 18 10:33:48] DEBUG[13112] stasis/app.c: Channel '212995' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13112] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Outgoing Call for 79821117045 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13114] chan_sip.c: Audio is at 17318 [Aug 18 10:33:48] VERBOSE[13114] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13114] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13114] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Initializing initreq for method INVITE - callid 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117045@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400108be [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13112] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 3 [ 52]: From: ;tag=as0d63cc42 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 6 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13114] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117045@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400108be Max-Forwards: 70 From: ;tag=as0d63cc42 To: Contact: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 892133707 892133707 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17318 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:48] DEBUG[13114] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13114] dial.c: Called zvonobot/79821117045 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400108be;received=159.65.48.104 From: ;tag=as0d63cc42 To: ;tag=as24c12118 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5cc7861d" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400108be;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0d63cc42 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as24c12118 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5cc7861d" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 (Checking To) --From tag as0d63cc42 --To-tag as24c12118 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117045@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK400108be Max-Forwards: 70 From: ;tag=as0d63cc42 To: ;tag=as24c12118 Contact: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 17318 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117045@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7cfa6980 Max-Forwards: 70 From: ;tag=as0d63cc42 To: Contact: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117045@178.62.121.41", nonce="5cc7861d", response="2a7bfc3651be30c8731dd991e30633d1" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 892133707 892133708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17318 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7cfa6980;received=159.65.48.104 From: ;tag=as0d63cc42 To: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7cfa6980;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0d63cc42 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 (Checking To) --From tag as0d63cc42 --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13117] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13117] http.c: HTTP Request URI is /ari/channels/212997?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117043&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13117] http.c: match request [ari/channels/212997] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13117] http.c: match request [ari/channels/212997] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13117] http.c: match request [ari/channels/212997] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13117] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13117] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Finding handler for channels/212997 [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Finding handler for 212997 [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking channels create: Didn't match 212997 [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13117] res_ari.c: Checking channels externalMedia: Didn't match 212997 [Aug 18 10:33:48] DEBUG[13117] res_ari.c: No explicit handler found for 212997. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: Allocating new SIP dialog for 4d07c5021094ccc8323db96a3c2d0b37@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13117] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c94011870' [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) RTP allocated port 10294 [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE creating session 0.0.0.0:10294 (10294) [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE create [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE add system candidates [Aug 18 10:33:48] DEBUG[13117] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13117] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE add candidate: 159.65.48.104:10294, 2130706431 [Aug 18 10:33:48] DEBUG[13117] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13117] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE add candidate: 10.131.0.10:10294, 2130706431 [Aug 18 10:33:48] DEBUG[13117] rtp_engine.c: RTP instance '0x7f0c94011870' is setup and ready to go [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) ICE stopped [Aug 18 10:33:48] DEBUG[13117] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13117] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13117] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13117] res_rtp_asterisk.c: (0x7f0c94011870) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13117] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13117] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: SIP call-id changed from '4d07c5021094ccc8323db96a3c2d0b37@127.0.1.1:5060' to '00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13117] stasis.c: Creating topic. name: channel:212997, detail: [Aug 18 10:33:48] DEBUG[13117] stasis.c: Topic 'channel:212997': 0x7f0c94018e10 created [Aug 18 10:33:48] DEBUG[13117] stasis.c: Creating topic. name: cache:49/channel:212997, detail: [Aug 18 10:33:48] DEBUG[13117] stasis.c: Topic 'cache:49/channel:212997': 0x7f0c94019010 created [Aug 18 10:33:48] DEBUG[13117] channel.c: Channel 0x7f0c94016e80 'SIP/zvonobot-00000021' allocated [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13117] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13117] res_stasis.c: calls_0: Subscribing to 212997 [Aug 18 10:33:48] DEBUG[13117] stasis/app.c: Channel '212997' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13117] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13117] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Outgoing Call for 79821117043 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13119] chan_sip.c: Audio is at 10294 [Aug 18 10:33:48] VERBOSE[13119] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13119] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13119] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Initializing initreq for method INVITE - callid 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117043@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK763fa955 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 3 [ 52]: From: ;tag=as0fc45651 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 6 [ 60]: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13119] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117043@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK763fa955 Max-Forwards: 70 From: ;tag=as0fc45651 To: Contact: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1108970894 1108970894 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10294 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:48] DEBUG[13119] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13119] dial.c: Called zvonobot/79821117043 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK763fa955;received=159.65.48.104 From: ;tag=as0fc45651 To: ;tag=as1bda8e63 Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08a8eb3a" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK763fa955;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0fc45651 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1bda8e63 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08a8eb3a" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 (Checking To) --From tag as0fc45651 --To-tag as1bda8e63 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117043@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK763fa955 Max-Forwards: 70 From: ;tag=as0fc45651 To: ;tag=as1bda8e63 Contact: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 10294 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117043@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ee695cb Max-Forwards: 70 From: ;tag=as0fc45651 To: Contact: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117043@178.62.121.41", nonce="08a8eb3a", response="8f24226ef8879fa32159726856deec40" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1108970894 1108970895 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10294 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ee695cb;received=159.65.48.104 From: ;tag=as0fc45651 To: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ee695cb;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0fc45651 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060 (Checking To) --From tag as0fc45651 --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '00ca721c392fff2043c05dfb2ae9e8e1@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13124] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13124] http.c: HTTP Request URI is /ari/channels/212998?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117042&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13124] http.c: match request [ari/channels/212998] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13124] http.c: match request [ari/channels/212998] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13124] http.c: match request [ari/channels/212998] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13124] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13124] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Finding handler for channels/212998 [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Finding handler for 212998 [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking channels create: Didn't match 212998 [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13124] res_ari.c: Checking channels externalMedia: Didn't match 212998 [Aug 18 10:33:48] DEBUG[13124] res_ari.c: No explicit handler found for 212998. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: Allocating new SIP dialog for 3288b51a21fb53017db578233812a9a5@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13124] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca800f800' [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) RTP allocated port 10026 [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE creating session 0.0.0.0:10026 (10026) [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE create [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE add system candidates [Aug 18 10:33:48] DEBUG[13124] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13124] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE add candidate: 159.65.48.104:10026, 2130706431 [Aug 18 10:33:48] DEBUG[13124] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13124] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE add candidate: 10.131.0.10:10026, 2130706431 [Aug 18 10:33:48] DEBUG[13124] rtp_engine.c: RTP instance '0x7f0ca800f800' is setup and ready to go [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) ICE stopped [Aug 18 10:33:48] DEBUG[13124] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13124] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13124] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13124] res_rtp_asterisk.c: (0x7f0ca800f800) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13124] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13124] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: SIP call-id changed from '3288b51a21fb53017db578233812a9a5@127.0.1.1:5060' to '26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13124] stasis.c: Creating topic. name: channel:212998, detail: [Aug 18 10:33:48] DEBUG[13124] stasis.c: Topic 'channel:212998': 0x7f0ca8016710 created [Aug 18 10:33:48] DEBUG[13124] stasis.c: Creating topic. name: cache:50/channel:212998, detail: [Aug 18 10:33:48] DEBUG[13124] stasis.c: Topic 'cache:50/channel:212998': 0x7f0ca80167f0 created [Aug 18 10:33:48] DEBUG[13124] channel.c: Channel 0x7f0ca80146e0 'SIP/zvonobot-00000022' allocated [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13124] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13124] res_stasis.c: calls_0: Subscribing to 212998 [Aug 18 10:33:48] DEBUG[13124] stasis/app.c: Channel '212998' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Outgoing Call for 79821117042 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13124] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13124] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13125] chan_sip.c: Audio is at 10026 [Aug 18 10:33:48] VERBOSE[13125] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13125] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13125] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Initializing initreq for method INVITE - callid 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117042@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK664212be [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 3 [ 52]: From: ;tag=as3cda4b3d [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 6 [ 60]: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13126] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13126] http.c: HTTP Request URI is /ari/channels/212999?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117041&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] DEBUG[13126] http.c: match request [ari/channels/212999] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13126] http.c: match request [ari/channels/212999] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13126] http.c: match request [ari/channels/212999] with handler [ari] len 3 [Aug 18 10:33:48] VERBOSE[13125] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117042@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK664212be Max-Forwards: 70 From: ;tag=as3cda4b3d To: Contact: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1250547912 1250547912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10026 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13126] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Aug 18 10:33:48] DEBUG[13126] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13125] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Finding handler for channels/212999 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Finding handler for channels [Aug 18 10:33:48] VERBOSE[13125] dial.c: Called zvonobot/79821117042 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK664212be;received=159.65.48.104 From: ;tag=as3cda4b3d To: ;tag=as2b45bf1a Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34223c87" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK664212be;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3cda4b3d [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2b45bf1a [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34223c87" [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 (Checking To) --From tag as3cda4b3d --To-tag as2b45bf1a [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117042@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK664212be Max-Forwards: 70 From: ;tag=as3cda4b3d To: ;tag=as2b45bf1a Contact: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 10026 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117042@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18 Max-Forwards: 70 From: ;tag=as3cda4b3d To: Contact: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117042@178.62.121.41", nonce="34223c87", response="50015fabc6e4b82e12272ac46bd2d40b" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1250547912 1250547913 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10026 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18;received=159.65.48.104 From: ;tag=as3cda4b3d To: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3cda4b3d [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 (Checking To) --From tag as3cda4b3d --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Finding handler for 212999 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking channels create: Didn't match 212999 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13126] res_ari.c: Checking channels externalMedia: Didn't match 212999 [Aug 18 10:33:48] DEBUG[13126] res_ari.c: No explicit handler found for 212999. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: Allocating new SIP dialog for 6da167797b8491503848346c236b2b76@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13126] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9800afc0' [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP allocated port 10060 [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE creating session 0.0.0.0:10060 (10060) [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE create [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE add system candidates [Aug 18 10:33:48] DEBUG[13126] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13126] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE add candidate: 159.65.48.104:10060, 2130706431 [Aug 18 10:33:48] DEBUG[13126] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13126] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE add candidate: 10.131.0.10:10060, 2130706431 [Aug 18 10:33:48] DEBUG[13126] rtp_engine.c: RTP instance '0x7f0c9800afc0' is setup and ready to go [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE stopped [Aug 18 10:33:48] DEBUG[13126] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13126] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13126] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13126] res_rtp_asterisk.c: (0x7f0c9800afc0) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13126] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13126] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: SIP call-id changed from '6da167797b8491503848346c236b2b76@127.0.1.1:5060' to '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13126] stasis.c: Creating topic. name: channel:212999, detail: [Aug 18 10:33:48] DEBUG[13126] stasis.c: Topic 'channel:212999': 0x7f0c980793e0 created [Aug 18 10:33:48] DEBUG[13126] stasis.c: Creating topic. name: cache:51/channel:212999, detail: [Aug 18 10:33:48] DEBUG[13126] stasis.c: Topic 'cache:51/channel:212999': 0x7f0c98078800 created [Aug 18 10:33:48] DEBUG[13126] channel.c: Channel 0x7f0c98012a90 'SIP/zvonobot-00000023' allocated [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13126] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13126] res_stasis.c: calls_0: Subscribing to 212999 [Aug 18 10:33:48] DEBUG[13126] stasis/app.c: Channel '212999' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Outgoing Call for 79821117041 [Aug 18 10:33:48] DEBUG[13126] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13129] chan_sip.c: Audio is at 10060 [Aug 18 10:33:48] VERBOSE[13129] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13129] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13129] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Initializing initreq for method INVITE - callid 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117041@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1646359c [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 3 [ 52]: From: ;tag=as3a1fc7ed [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 6 [ 60]: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13129] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117041@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1646359c Max-Forwards: 70 From: ;tag=as3a1fc7ed To: Contact: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2133221329 2133221329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10060 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:48] DEBUG[13129] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[13126] http.c: HTTP closing session. Top level [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1646359c;received=159.65.48.104 From: ;tag=as3a1fc7ed To: ;tag=as02189fdc Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22eff029" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1646359c;received=159.65.48.104 [Aug 18 10:33:48] VERBOSE[13129] dial.c: Called zvonobot/79821117041 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a1fc7ed [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as02189fdc [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22eff029" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 (Checking To) --From tag as3a1fc7ed --To-tag as02189fdc [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117041@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1646359c Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as02189fdc Contact: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 10060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117041@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f2da01 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: Contact: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41", nonce="22eff029", response="e8ec64b7ab28b8686f90393d4bc59149" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2133221329 2133221330 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10060 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[13131] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13131] http.c: HTTP Request URI is /ari/channels/213000?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117040&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13131] http.c: match request [ari/channels/213000] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13131] http.c: match request [ari/channels/213000] with handler [phoneprov] len 9 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f2da01;received=159.65.48.104 From: ;tag=as3a1fc7ed To: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f2da01;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a1fc7ed [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 (Checking To) --From tag as3a1fc7ed --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13131] http.c: match request [ari/channels/213000] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13131] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13131] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Finding handler for channels/213000 [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Finding handler for 213000 [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking channels create: Didn't match 213000 [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13131] res_ari.c: Checking channels externalMedia: Didn't match 213000 [Aug 18 10:33:48] DEBUG[13131] res_ari.c: No explicit handler found for 213000. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: Allocating new SIP dialog for 1d210f423af8fab6307a712367ddf19d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13131] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca401ab20' [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP allocated port 12158 [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE creating session 0.0.0.0:12158 (12158) [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE create [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE add system candidates [Aug 18 10:33:48] DEBUG[13131] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13131] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE add candidate: 159.65.48.104:12158, 2130706431 [Aug 18 10:33:48] DEBUG[13131] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13131] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE add candidate: 10.131.0.10:12158, 2130706431 [Aug 18 10:33:48] DEBUG[13131] rtp_engine.c: RTP instance '0x7f0ca401ab20' is setup and ready to go [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE stopped [Aug 18 10:33:48] DEBUG[13131] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13131] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13131] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13131] res_rtp_asterisk.c: (0x7f0ca401ab20) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13131] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13131] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: SIP call-id changed from '1d210f423af8fab6307a712367ddf19d@127.0.1.1:5060' to '2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13131] stasis.c: Creating topic. name: channel:213000, detail: [Aug 18 10:33:48] DEBUG[13131] stasis.c: Topic 'channel:213000': 0x7f0ca4024390 created [Aug 18 10:33:48] DEBUG[13131] stasis.c: Creating topic. name: cache:52/channel:213000, detail: [Aug 18 10:33:48] DEBUG[13131] stasis.c: Topic 'cache:52/channel:213000': 0x7f0ca40224a0 created [Aug 18 10:33:48] DEBUG[13131] channel.c: Channel 0x7f0ca4022610 'SIP/zvonobot-00000024' allocated [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13131] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13131] res_stasis.c: calls_0: Subscribing to 213000 [Aug 18 10:33:48] DEBUG[13131] stasis/app.c: Channel '213000' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Outgoing Call for 79821117040 [Aug 18 10:33:48] DEBUG[13131] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13132] chan_sip.c: Audio is at 12158 [Aug 18 10:33:48] VERBOSE[13132] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13132] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13132] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Initializing initreq for method INVITE - callid 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117040@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16cb800f [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 3 [ 52]: From: ;tag=as4406e1db [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 6 [ 60]: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13131] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13132] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117040@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16cb800f Max-Forwards: 70 From: ;tag=as4406e1db To: Contact: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1311716036 1311716036 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:48] DEBUG[13132] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13132] dial.c: Called zvonobot/79821117040 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16cb800f;received=159.65.48.104 From: ;tag=as4406e1db To: ;tag=as3c7ae2c0 Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0411341c" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16cb800f;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4406e1db [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3c7ae2c0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0411341c" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 (Checking To) --From tag as4406e1db --To-tag as3c7ae2c0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117040@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16cb800f Max-Forwards: 70 From: ;tag=as4406e1db To: ;tag=as3c7ae2c0 Contact: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 12158 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117040@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31f6c45a Max-Forwards: 70 From: ;tag=as4406e1db To: Contact: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117040@178.62.121.41", nonce="0411341c", response="937d0d4b953a95c8127b94f8cfd1e594" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1311716036 1311716037 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31f6c45a;received=159.65.48.104 From: ;tag=as4406e1db To: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31f6c45a;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4406e1db [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 (Checking To) --From tag as4406e1db --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #2 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13134] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13134] http.c: HTTP Request URI is /ari/channels/213001?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117039&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13134] http.c: match request [ari/channels/213001] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13134] http.c: match request [ari/channels/213001] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13134] http.c: match request [ari/channels/213001] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13134] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13134] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Finding handler for channels/213001 [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Finding handler for 213001 [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking channels create: Didn't match 213001 [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13134] res_ari.c: Checking channels externalMedia: Didn't match 213001 [Aug 18 10:33:48] DEBUG[13134] res_ari.c: No explicit handler found for 213001. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: Allocating new SIP dialog for 4c796a053dcce3aa489bf5aa3bc16947@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13134] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac020760' [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) RTP allocated port 12390 [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE creating session 0.0.0.0:12390 (12390) [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE create [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE add system candidates [Aug 18 10:33:48] DEBUG[13134] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13134] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE add candidate: 159.65.48.104:12390, 2130706431 [Aug 18 10:33:48] DEBUG[13134] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13134] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE add candidate: 10.131.0.10:12390, 2130706431 [Aug 18 10:33:48] DEBUG[13134] rtp_engine.c: RTP instance '0x7f0cac020760' is setup and ready to go [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) ICE stopped [Aug 18 10:33:48] DEBUG[13134] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13134] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13134] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13134] res_rtp_asterisk.c: (0x7f0cac020760) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13134] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13134] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: SIP call-id changed from '4c796a053dcce3aa489bf5aa3bc16947@127.0.1.1:5060' to '58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13134] stasis.c: Creating topic. name: channel:213001, detail: [Aug 18 10:33:48] DEBUG[13134] stasis.c: Topic 'channel:213001': 0x7f0cac0277a0 created [Aug 18 10:33:48] DEBUG[13134] stasis.c: Creating topic. name: cache:53/channel:213001, detail: [Aug 18 10:33:48] DEBUG[13134] stasis.c: Topic 'cache:53/channel:213001': 0x7f0cac08c250 created [Aug 18 10:33:48] DEBUG[13134] channel.c: Channel 0x7f0cac025a20 'SIP/zvonobot-00000025' allocated [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13134] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13134] res_stasis.c: calls_0: Subscribing to 213001 [Aug 18 10:33:48] DEBUG[13134] stasis/app.c: Channel '213001' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Outgoing Call for 79821117039 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13134] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13135] chan_sip.c: Audio is at 12390 [Aug 18 10:33:48] VERBOSE[13135] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13135] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13135] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Initializing initreq for method INVITE - callid 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117039@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK136656bb [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 3 [ 52]: From: ;tag=as03e5279c [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 6 [ 60]: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13135] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117039@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK136656bb Max-Forwards: 70 From: ;tag=as03e5279c To: Contact: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 919879013 919879013 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:48] DEBUG[13135] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] VERBOSE[13135] dial.c: Called zvonobot/79821117039 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK136656bb;received=159.65.48.104 From: ;tag=as03e5279c To: ;tag=as623a545e Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="356ceb14" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK136656bb;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as03e5279c [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as623a545e [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="356ceb14" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 (Checking To) --From tag as03e5279c --To-tag as623a545e [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117039@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK136656bb Max-Forwards: 70 From: ;tag=as03e5279c To: ;tag=as623a545e Contact: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 12390 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117039@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK775ec7d3 Max-Forwards: 70 From: ;tag=as03e5279c To: Contact: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117039@178.62.121.41", nonce="356ceb14", response="c665c83cf3b5b49f50c52cad98223a55" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 919879013 919879014 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[13134] http.c: HTTP closing session. Top level [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK775ec7d3;received=159.65.48.104 From: ;tag=as03e5279c To: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK775ec7d3;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as03e5279c [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060 (Checking To) --From tag as03e5279c --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '58a7d3c9780820995ada78d40ded521e@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13137] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13137] http.c: HTTP Request URI is /ari/channels/213003?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117037&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13137] http.c: match request [ari/channels/213003] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13137] http.c: match request [ari/channels/213003] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13137] http.c: match request [ari/channels/213003] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13137] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13137] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Finding handler for channels/213003 [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Finding handler for 213003 [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking channels create: Didn't match 213003 [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13137] res_ari.c: Checking channels externalMedia: Didn't match 213003 [Aug 18 10:33:48] DEBUG[13137] res_ari.c: No explicit handler found for 213003. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: Allocating new SIP dialog for 73adea1a5381a6f339c9c81106930f2a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13137] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c14110' [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) RTP allocated port 19316 [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE creating session 0.0.0.0:19316 (19316) [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE create [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE add system candidates [Aug 18 10:33:48] DEBUG[13137] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13137] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE add candidate: 159.65.48.104:19316, 2130706431 [Aug 18 10:33:48] DEBUG[13137] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13137] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE add candidate: 10.131.0.10:19316, 2130706431 [Aug 18 10:33:48] DEBUG[13137] rtp_engine.c: RTP instance '0x2c14110' is setup and ready to go [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) ICE stopped [Aug 18 10:33:48] DEBUG[13137] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13137] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13137] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13137] res_rtp_asterisk.c: (0x2c14110) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13137] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13137] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: SIP call-id changed from '73adea1a5381a6f339c9c81106930f2a@127.0.1.1:5060' to '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13137] stasis.c: Creating topic. name: channel:213003, detail: [Aug 18 10:33:48] DEBUG[13137] stasis.c: Topic 'channel:213003': 0x2c1d880 created [Aug 18 10:33:48] DEBUG[13137] stasis.c: Creating topic. name: cache:54/channel:213003, detail: [Aug 18 10:33:48] DEBUG[13137] stasis.c: Topic 'cache:54/channel:213003': 0x2c81730 created [Aug 18 10:33:48] DEBUG[13137] channel.c: Channel 0x2c1bb30 'SIP/zvonobot-00000026' allocated [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13137] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13137] res_stasis.c: calls_0: Subscribing to 213003 [Aug 18 10:33:48] DEBUG[13137] stasis/app.c: Channel '213003' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Outgoing Call for 79821117037 [Aug 18 10:33:48] DEBUG[13137] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13137] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13138] chan_sip.c: Audio is at 19316 [Aug 18 10:33:48] VERBOSE[13138] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13138] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13138] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Initializing initreq for method INVITE - callid 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117037@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a1c58d3 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 3 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 6 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13138] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117037@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a1c58d3 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1994638417 1994638417 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:48] DEBUG[13138] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[13138] dial.c: Called zvonobot/79821117037 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a1c58d3;received=159.65.48.104 From: ;tag=as4ca0b4bd To: ;tag=as3f3fe3b9 Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3031e556" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a1c58d3;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3f3fe3b9 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3031e556" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 (Checking To) --From tag as4ca0b4bd --To-tag as3f3fe3b9 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117037@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a1c58d3 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: ;tag=as3f3fe3b9 Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 19316 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117037@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117037@178.62.121.41", nonce="3031e556", response="31d658c7e448bb09a6f9272de6c9701d" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1994638417 1994638418 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 From: ;tag=as4ca0b4bd To: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 (Checking To) --From tag as4ca0b4bd --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] DEBUG[13140] http.c: HTTP opening session. Top level [Aug 18 10:33:48] DEBUG[13140] http.c: HTTP Request URI is /ari/channels/213002?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117038&callerId=74950493843 [Aug 18 10:33:48] DEBUG[13140] http.c: match request [ari/channels/213002] with handler [httpstatus] len 10 [Aug 18 10:33:48] DEBUG[13140] http.c: match request [ari/channels/213002] with handler [phoneprov] len 9 [Aug 18 10:33:48] DEBUG[13140] http.c: match request [ari/channels/213002] with handler [ari] len 3 [Aug 18 10:33:48] DEBUG[13140] http.c: Match made with [ari] [Aug 18 10:33:48] DEBUG[13140] http.c: HTTP consuming request body [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Finding handler for channels/213002 [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Finding handler for channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Finding handler for 213002 [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking channels create: Didn't match 213002 [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:48] DEBUG[13140] res_ari.c: Checking channels externalMedia: Didn't match 213002 [Aug 18 10:33:48] DEBUG[13140] res_ari.c: No explicit handler found for 213002. Using wildcard channelId. [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: Allocating new SIP dialog for 59c9d5ad2f00bef92c8071dd41645cb6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:48] DEBUG[13140] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c0801b610' [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) RTP allocated port 15074 [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE creating session 0.0.0.0:15074 (15074) [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE create [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE add system candidates [Aug 18 10:33:48] DEBUG[13140] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:48] DEBUG[13140] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE add candidate: 159.65.48.104:15074, 2130706431 [Aug 18 10:33:48] DEBUG[13140] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:48] DEBUG[13140] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE add candidate: 10.131.0.10:15074, 2130706431 [Aug 18 10:33:48] DEBUG[13140] rtp_engine.c: RTP instance '0x7f0c0801b610' is setup and ready to go [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) ICE stopped [Aug 18 10:33:48] DEBUG[13140] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:48] DEBUG[13140] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:48] DEBUG[13140] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:48] DEBUG[13140] res_rtp_asterisk.c: (0x7f0c0801b610) RTCP setup on RTP instance [Aug 18 10:33:48] VERBOSE[13140] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:48] DEBUG[13140] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: SIP call-id changed from '59c9d5ad2f00bef92c8071dd41645cb6@127.0.1.1:5060' to '25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060' [Aug 18 10:33:48] DEBUG[13140] stasis.c: Creating topic. name: channel:213002, detail: [Aug 18 10:33:48] DEBUG[13140] stasis.c: Topic 'channel:213002': 0x7f0c08022f50 created [Aug 18 10:33:48] DEBUG[13140] stasis.c: Creating topic. name: cache:55/channel:213002, detail: [Aug 18 10:33:48] DEBUG[13140] stasis.c: Topic 'cache:55/channel:213002': 0x7f0c08088c40 created [Aug 18 10:33:48] DEBUG[13140] channel.c: Channel 0x7f0c08023460 'SIP/zvonobot-00000027' allocated [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:48] DEBUG[13140] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:48] DEBUG[13140] res_stasis.c: calls_0: Subscribing to 213002 [Aug 18 10:33:48] DEBUG[13140] stasis/app.c: Channel '213002' is 1 interested in calls_0 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Outgoing Call for 79821117038 [Aug 18 10:33:48] DEBUG[13140] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:48] DEBUG[13140] http.c: HTTP closing session. Top level [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[13141] chan_sip.c: Audio is at 15074 [Aug 18 10:33:48] VERBOSE[13141] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[13141] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[13141] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Initializing initreq for method INVITE - callid 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117038@178.62.121.41 SIP/2.0 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0f6188c0 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 3 [ 52]: From: ;tag=as5d2e4a10 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 6 [ 60]: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:48 GMT [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:48] VERBOSE[13141] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117038@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0f6188c0 Max-Forwards: 70 From: ;tag=as5d2e4a10 To: Contact: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1070580995 1070580995 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15074 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:48] DEBUG[13141] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:48] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:48] VERBOSE[13141] dial.c: Called zvonobot/79821117038 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0f6188c0;received=159.65.48.104 From: ;tag=as5d2e4a10 To: ;tag=as7a0d4a1b Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="24cd596d" Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0f6188c0;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5d2e4a10 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7a0d4a1b [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="24cd596d" [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 (Checking To) --From tag as5d2e4a10 --To-tag as7a0d4a1b [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Stopping retransmission on '25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117038@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0f6188c0 Max-Forwards: 70 From: ;tag=as5d2e4a10 To: ;tag=as7a0d4a1b Contact: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Audio is at 15074 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117038@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4be660db Max-Forwards: 70 From: ;tag=as5d2e4a10 To: Contact: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117038@178.62.121.41", nonce="24cd596d", response="0d6b329fb9321eaccacc18d6984ab32f" Date: Wed, 18 Aug 2021 10:33:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1070580995 1070580996 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15074 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4be660db;received=159.65.48.104 From: ;tag=as5d2e4a10 To: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4be660db;received=159.65.48.104 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5d2e4a10 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:48] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 (Checking To) --From tag as5d2e4a10 --To-tag [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:48] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:48] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:33:48] VERBOSE[13148] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:33:49] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca;received=159.65.48.104 From: ;tag=as0aff19ec To: ;tag=as0ddec481 Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca;received=159.65.48.104 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0aff19ec [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0ddec481 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:49] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 (Checking To) --From tag as0aff19ec --To-tag as0ddec481 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Stopping retransmission on '36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:49] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:33:49] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:49] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:49] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117070@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK686be5ca Max-Forwards: 70 From: ;tag=as0aff19ec To: ;tag=as0ddec481 Contact: Call-ID: 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:49] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:49] VERBOSE[12891] dial.c: SIP/zvonobot-00000005 is busy [Aug 18 10:33:49] DEBUG[12891] channel.c: Channel 0x7f0c2c010ca0 'SIP/zvonobot-00000005' hanging up. Refs: 2 [Aug 18 10:33:49] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000005 - start 1629282822.162763 answer 0.000000 end 1629282829.176411 dur 7.013 bill 1629282829.176 dispo BUSY [Aug 18 10:33:49] DEBUG[12891] chan_sip.c: Hangup call SIP/zvonobot-00000005, SIP callid 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:49] DEBUG[12891] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:49] DEBUG[12891] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:49] DEBUG[12891] channel.c: Channel 0x7f0c2c010ca0 'SIP/zvonobot-00000005' destroying [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:49] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282829.47, detail: [Aug 18 10:33:49] DEBUG[20545] stasis.c: Topic 'channel:1629282829.47': 0x7f0c300483a0 created [Aug 18 10:33:49] DEBUG[20545] stasis.c: Creating topic. name: cache:56/channel:1629282829.47, detail: [Aug 18 10:33:49] DEBUG[20545] stasis.c: Topic 'cache:56/channel:1629282829.47': 0x7f0c30048d70 created [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:33:49] DEBUG[20620] stasis/app.c: channel '212970': is 0 interested in calls_0 [Aug 18 10:33:49] DEBUG[20620] stasis/app.c: channel '212970' unsubscribed from calls_0 [Aug 18 10:33:49] DEBUG[20545] stasis.c: Destroying topic. name: cache:56/channel:1629282829.47, detail: [Aug 18 10:33:49] DEBUG[20545] stasis.c: Topic 'cache:56/channel:1629282829.47': 0x7f0c30048d70 destroyed [Aug 18 10:33:49] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282829.47, detail: [Aug 18 10:33:49] DEBUG[20545] stasis.c: Topic 'channel:1629282829.47': 0x7f0c300483a0 destroyed [Aug 18 10:33:49] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000005', '', 'AppDial2', '(Outgoing Line)', 7, 0, 'BUSY', 3, '', '212970', '')] [Aug 18 10:33:49] DEBUG[13149] http.c: HTTP opening session. Top level [Aug 18 10:33:49] DEBUG[13149] http.c: HTTP Request URI is /ari/channels/212970 [Aug 18 10:33:49] DEBUG[13149] http.c: match request [ari/channels/212970] with handler [httpstatus] len 10 [Aug 18 10:33:49] DEBUG[13149] http.c: match request [ari/channels/212970] with handler [phoneprov] len 9 [Aug 18 10:33:49] DEBUG[13149] http.c: match request [ari/channels/212970] with handler [ari] len 3 [Aug 18 10:33:49] DEBUG[13149] http.c: Match made with [ari] [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Finding handler for channels/212970 [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Finding handler for channels [Aug 18 10:33:49] DEBUG[12891] stasis.c: Destroying topic. name: cache:12/channel:212970, detail: [Aug 18 10:33:49] DEBUG[12891] stasis.c: Topic 'cache:12/channel:212970': 0x7f0c2c012960 destroyed [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:49] DEBUG[12891] stasis.c: Destroying topic. name: channel:212970, detail: [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:49] DEBUG[12891] stasis.c: Topic 'channel:212970': 0x7f0c2c012760 destroyed [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Finding handler for 212970 [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking channels create: Didn't match 212970 [Aug 18 10:33:49] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:49] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:49] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:49] DEBUG[13149] res_ari.c: Checking channels externalMedia: Didn't match 212970 [Aug 18 10:33:49] DEBUG[13149] res_ari.c: No explicit handler found for 212970. Using wildcard channelId. [Aug 18 10:33:49] DEBUG[13149] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:49] DEBUG[13149] http.c: HTTP closing session. Top level [Aug 18 10:33:49] DEBUG[13075] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:49] DEBUG[13075] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:49] DEBUG[13075] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:49] DEBUG[13075] channel.c: Channel Announcer/ARI-00000001;1 setting write format path: slin -> slin [Aug 18 10:33:49] DEBUG[13075] channel.c: Channel 0x7f0c2402b670 'Announcer/ARI-00000001;1' hanging up. Refs: 2 [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:49] DEBUG[13075] channel.c: Channel 0x7f0c2402b670 'Announcer/ARI-00000001;1' destroying [Aug 18 10:33:49] DEBUG[13071] bridge_channel.c: Setting 0x7f0c240386d0(Announcer/ARI-00000001;2) state from:0 to:1 [Aug 18 10:33:49] DEBUG[13071] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pulling 0x7f0c240386d0(Announcer/ARI-00000001;2) [Aug 18 10:33:49] VERBOSE[13071] bridge_channel.c: Channel Announcer/ARI-00000001;2 left 'softmix' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:33:49] DEBUG[13071] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c240386d0(Announcer/ARI-00000001;2) is leaving softmix technology [Aug 18 10:33:49] DEBUG[13075] stasis.c: Destroying topic. name: cache:43/channel:1629282827.34, detail: [Aug 18 10:33:49] DEBUG[13150] http.c: HTTP opening session. Top level [Aug 18 10:33:49] DEBUG[13075] stasis.c: Topic 'cache:43/channel:1629282827.34': 0x7f0c240089d0 destroyed [Aug 18 10:33:49] DEBUG[13150] http.c: HTTP Request URI is /ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:49] DEBUG[13150] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [httpstatus] len 10 [Aug 18 10:33:49] DEBUG[13150] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [phoneprov] len 9 [Aug 18 10:33:49] DEBUG[13071] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c'. Checking compatability for channels 'SIP/zvonobot-00000000' and 'Recorder/ARI-00000000;2' [Aug 18 10:33:49] DEBUG[13150] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play] with handler [ari] len 3 [Aug 18 10:33:49] DEBUG[13075] stasis.c: Destroying topic. name: channel:1629282827.34, detail: [Aug 18 10:33:49] DEBUG[13150] http.c: Match made with [ari] [Aug 18 10:33:49] DEBUG[13071] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as channel 'SIP/zvonobot-00000000' has features which prevent it [Aug 18 10:33:49] DEBUG[13075] stasis.c: Topic 'channel:1629282827.34': 0x7f0c24006840 destroyed [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:49] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:49] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:49] DEBUG[13071] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:49] VERBOSE[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: switching from softmix technology to simple_bridge [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology constructor [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Finding handler for bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c/play [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Finding handler for bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0cac00a6f0(SIP/zvonobot-00000000) to dummy bridge temporarily [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0ca400f4f0(Recorder/ARI-00000000;2) to dummy bridge temporarily [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is leaving softmix technology (dummy) [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is leaving softmix technology (dummy) [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology stop [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining simple_bridge technology [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Finding handler for 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel Recorder/ARI-00000000;2 setting read format path: slin -> slin [Aug 18 10:33:49] DEBUG[13150] res_ari.c: No explicit handler found for 7421ba4f-6229-4eeb-b806-91ebc84ff38c. Using wildcard bridgeId. [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Finding handler for play [Aug 18 10:33:49] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:49] DEBUG[13150] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:33:49] DEBUG[13150] stasis.c: Creating topic. name: channel:1629282829.48, detail: [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining simple_bridge technology [Aug 18 10:33:49] DEBUG[13150] stasis.c: Topic 'channel:1629282829.48': 0x7f0c20010a40 created [Aug 18 10:33:49] DEBUG[13150] stasis.c: Creating topic. name: cache:57/channel:1629282829.48, detail: [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel Recorder/ARI-00000000;2 setting read format path: slin -> slin [Aug 18 10:33:49] DEBUG[13150] stasis.c: Topic 'cache:57/channel:1629282829.48': 0x7f0c20033060 created [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology start [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: deferring softmix technology destructor [Aug 18 10:33:49] DEBUG[13071] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: queueing action type:13 sub:1000 [Aug 18 10:33:49] DEBUG[13150] channel.c: Channel 0x7f0c20014960 'Announcer/ARI-00000002;1' allocated [Aug 18 10:33:49] DEBUG[13150] stasis.c: Creating topic. name: channel:1629282829.49, detail: [Aug 18 10:33:49] DEBUG[13150] stasis.c: Topic 'channel:1629282829.49': 0x7f0c2000d150 created [Aug 18 10:33:49] DEBUG[13150] stasis.c: Creating topic. name: cache:58/channel:1629282829.49, detail: [Aug 18 10:33:49] DEBUG[13150] stasis.c: Topic 'cache:58/channel:1629282829.49': 0x7f0c2000cbc0 created [Aug 18 10:33:49] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:49] DEBUG[20534] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: Waiting for mixing thread to die. [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel 0x7f0c24031840 'Announcer/ARI-00000001;2' hanging up. Refs: 2 [Aug 18 10:33:49] DEBUG[13058] channel.c: Recorder/ARI-00000000;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:49] DEBUG[13071] channel.c: Channel 0x7f0c24031840 'Announcer/ARI-00000001;2' destroying [Aug 18 10:33:49] DEBUG[13056] channel.c: SIP/zvonobot-00000000: Dropping redundant connected line update "" <>. [Aug 18 10:33:49] DEBUG[13150] channel.c: Channel 0x7f0c2001ad30 'Announcer/ARI-00000002;2' allocated [Aug 18 10:33:49] DEBUG[13150] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:49] DEBUG[13071] stasis.c: Destroying topic. name: cache:45/channel:1629282827.36, detail: [Aug 18 10:33:49] DEBUG[13071] stasis.c: Topic 'cache:45/channel:1629282827.36': 0x7f0c24033210 destroyed [Aug 18 10:33:49] DEBUG[13150] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000002;1' [Aug 18 10:33:49] DEBUG[13071] stasis.c: Destroying topic. name: channel:1629282827.36, detail: [Aug 18 10:33:49] DEBUG[13071] stasis.c: Topic 'channel:1629282827.36': 0x7f0c2402cbf0 destroyed [Aug 18 10:33:49] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:49] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:49] DEBUG[13151] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c2001ab20(Announcer/ARI-00000002;2) is joining [Aug 18 10:33:49] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:49] DEBUG[13151] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pushing 0x7f0c2001ab20(Announcer/ARI-00000002;2) [Aug 18 10:33:49] DEBUG[13151] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:49] VERBOSE[13151] bridge_channel.c: Channel Announcer/ARI-00000002;2 joined 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:49] DEBUG[13151] bridge.c: Chose bridge technology softmix [Aug 18 10:33:49] VERBOSE[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: switching from simple_bridge technology to softmix [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology constructor [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0cac00a6f0(SIP/zvonobot-00000000) to dummy bridge temporarily [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0ca400f4f0(Recorder/ARI-00000000;2) to dummy bridge temporarily [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is leaving simple_bridge technology (dummy) [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology stop [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c2001ab20(Announcer/ARI-00000002;2) is joining softmix technology [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Announcer/ARI-00000002;2: [Aug 18 10:33:49] DEBUG[13151] channel.c: Channel Announcer/ARI-00000002;2 setting write format path: slin -> slin [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Announcer/ARI-00000002;2: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Announcer/ARI-00000002;2: Not in SFU mode [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining softmix technology [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: SIP/zvonobot-00000000: [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: SIP/zvonobot-00000000: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: SIP/zvonobot-00000000: Not in SFU mode [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining softmix technology [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Recorder/ARI-00000000;2: [Aug 18 10:33:49] DEBUG[13151] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:49] DEBUG[13151] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Recorder/ARI-00000000;2: [Aug 18 10:33:49] DEBUG[13151] bridge_softmix.c: Recorder/ARI-00000000;2: Not in SFU mode [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology start [Aug 18 10:33:49] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology destructor [Aug 18 10:33:49] DEBUG[13152] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: starting mixing thread [Aug 18 10:33:49] DEBUG[13150] res_stasis_playback.c: 1629282829.48: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:49] DEBUG[13153] channel.c: Channel Announcer/ARI-00000002;1 setting write format path: gsm -> slin [Aug 18 10:33:49] DEBUG[13153] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:49] VERBOSE[13153] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:49] DEBUG[13073] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: stopping mixing thread [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:49] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:49] DEBUG[13150] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:49] DEBUG[13150] http.c: HTTP closing session. Top level [Aug 18 10:33:49] DEBUG[20564] res_pjsip_registrar.c: Woke up at 1629282829 Interval: 30 [Aug 18 10:33:49] DEBUG[20564] res_pjsip_registrar.c: Expiring 0 contacts [Aug 18 10:33:49] DEBUG[13166] http.c: HTTP opening session. Top level [Aug 18 10:33:49] DEBUG[13166] http.c: HTTP Request URI is /ari/channels/213004?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117036&callerId=74950493843 [Aug 18 10:33:49] DEBUG[13166] http.c: match request [ari/channels/213004] with handler [httpstatus] len 10 [Aug 18 10:33:49] DEBUG[13166] http.c: match request [ari/channels/213004] with handler [phoneprov] len 9 [Aug 18 10:33:49] DEBUG[13166] http.c: match request [ari/channels/213004] with handler [ari] len 3 [Aug 18 10:33:49] DEBUG[13166] http.c: Match made with [ari] [Aug 18 10:33:49] DEBUG[13166] http.c: HTTP consuming request body [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Finding handler for channels/213004 [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Finding handler for channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Finding handler for 213004 [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking channels create: Didn't match 213004 [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:49] DEBUG[13166] res_ari.c: Checking channels externalMedia: Didn't match 213004 [Aug 18 10:33:49] DEBUG[13166] res_ari.c: No explicit handler found for 213004. Using wildcard channelId. [Aug 18 10:33:49] DEBUG[13166] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:49] DEBUG[13166] chan_sip.c: Allocating new SIP dialog for 3b8dbe7a68e00a9a68f3a6f059d86a1a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:49] DEBUG[13166] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c30031690' [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) RTP allocated port 18974 [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE creating session 0.0.0.0:18974 (18974) [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE create [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE add system candidates [Aug 18 10:33:50] DEBUG[13166] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13166] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE add candidate: 159.65.48.104:18974, 2130706431 [Aug 18 10:33:50] DEBUG[13166] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13166] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE add candidate: 10.131.0.10:18974, 2130706431 [Aug 18 10:33:50] DEBUG[13166] rtp_engine.c: RTP instance '0x7f0c30031690' is setup and ready to go [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) ICE stopped [Aug 18 10:33:50] DEBUG[13166] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13166] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13166] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13166] res_rtp_asterisk.c: (0x7f0c30031690) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13166] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13166] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: SIP call-id changed from '3b8dbe7a68e00a9a68f3a6f059d86a1a@127.0.1.1:5060' to '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13166] stasis.c: Creating topic. name: channel:213004, detail: [Aug 18 10:33:50] DEBUG[13166] stasis.c: Topic 'channel:213004': 0x7f0c300ab380 created [Aug 18 10:33:50] DEBUG[13166] stasis.c: Creating topic. name: cache:59/channel:213004, detail: [Aug 18 10:33:50] DEBUG[13166] stasis.c: Topic 'cache:59/channel:213004': 0x7f0c300ab560 created [Aug 18 10:33:50] DEBUG[13166] channel.c: Channel 0x7f0c30038fd0 'SIP/zvonobot-00000028' allocated [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13166] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13166] res_stasis.c: calls_0: Subscribing to 213004 [Aug 18 10:33:50] DEBUG[13166] stasis/app.c: Channel '213004' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Destroying SIP dialog 36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '36f65e526a5d1f2118078b4c6c374003@159.65.48.104:5060' Method: INVITE [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS stop [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c00bbb0) ICE RTP transport deallocating [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c00bbb0' [Aug 18 10:33:50] DEBUG[13166] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Outgoing Call for 79821117036 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13166] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13167] chan_sip.c: Audio is at 18974 [Aug 18 10:33:50] VERBOSE[13167] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13167] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13167] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Initializing initreq for method INVITE - callid 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117036@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06e20461 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 3 [ 52]: From: ;tag=as04f0121c [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 6 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13167] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06e20461 Max-Forwards: 70 From: ;tag=as04f0121c To: Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1956738346 1956738346 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18974 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:50] DEBUG[13167] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13167] dial.c: Called zvonobot/79821117036 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06e20461;received=159.65.48.104 From: ;tag=as04f0121c To: ;tag=as238c3367 Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7094f15a" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06e20461;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as04f0121c [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as238c3367 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7094f15a" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 (Checking To) --From tag as04f0121c --To-tag as238c3367 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06e20461 Max-Forwards: 70 From: ;tag=as04f0121c To: ;tag=as238c3367 Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 18974 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae Max-Forwards: 70 From: ;tag=as04f0121c To: Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117036@178.62.121.41", nonce="7094f15a", response="6c535b60228d27a4a4971e5aff2ec011" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1956738346 1956738347 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18974 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 From: ;tag=as04f0121c To: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as04f0121c [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 (Checking To) --From tag as04f0121c --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13170] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13170] http.c: HTTP Request URI is /ari/channels/213006?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117034&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13170] http.c: match request [ari/channels/213006] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13170] http.c: match request [ari/channels/213006] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13170] http.c: match request [ari/channels/213006] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13170] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13170] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Finding handler for channels/213006 [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Finding handler for 213006 [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking channels create: Didn't match 213006 [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13170] res_ari.c: Checking channels externalMedia: Didn't match 213006 [Aug 18 10:33:50] DEBUG[13170] res_ari.c: No explicit handler found for 213006. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: Allocating new SIP dialog for 5551d15f6b5d0e49479b38105ae3d0b9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13170] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38023850' [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) RTP allocated port 10586 [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE creating session 0.0.0.0:10586 (10586) [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE create [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE add system candidates [Aug 18 10:33:50] DEBUG[13170] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13170] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE add candidate: 159.65.48.104:10586, 2130706431 [Aug 18 10:33:50] DEBUG[13170] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13170] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE add candidate: 10.131.0.10:10586, 2130706431 [Aug 18 10:33:50] DEBUG[13170] rtp_engine.c: RTP instance '0x7f0c38023850' is setup and ready to go [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) ICE stopped [Aug 18 10:33:50] DEBUG[13170] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13170] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13170] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13170] res_rtp_asterisk.c: (0x7f0c38023850) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13170] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13170] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: SIP call-id changed from '5551d15f6b5d0e49479b38105ae3d0b9@127.0.1.1:5060' to '15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13170] stasis.c: Creating topic. name: channel:213006, detail: [Aug 18 10:33:50] DEBUG[13170] stasis.c: Topic 'channel:213006': 0x7f0c380924b0 created [Aug 18 10:33:50] DEBUG[13170] stasis.c: Creating topic. name: cache:60/channel:213006, detail: [Aug 18 10:33:50] DEBUG[13170] stasis.c: Topic 'cache:60/channel:213006': 0x7f0c3802d990 created [Aug 18 10:33:50] DEBUG[13170] channel.c: Channel 0x7f0c3802c5d0 'SIP/zvonobot-00000029' allocated [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13170] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13170] res_stasis.c: calls_0: Subscribing to 213006 [Aug 18 10:33:50] DEBUG[13170] stasis/app.c: Channel '213006' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13170] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13170] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Outgoing Call for 79821117034 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13171] chan_sip.c: Audio is at 10586 [Aug 18 10:33:50] VERBOSE[13171] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13171] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13171] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Initializing initreq for method INVITE - callid 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117034@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK167dd99b [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 3 [ 52]: From: ;tag=as7cfb6ac0 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 6 [ 60]: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13171] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117034@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK167dd99b Max-Forwards: 70 From: ;tag=as7cfb6ac0 To: Contact: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637470277 637470277 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:50] DEBUG[13171] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[13171] dial.c: Called zvonobot/79821117034 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK167dd99b;received=159.65.48.104 From: ;tag=as7cfb6ac0 To: ;tag=as1df42b05 Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="225462f9" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK167dd99b;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7cfb6ac0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1df42b05 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="225462f9" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 (Checking To) --From tag as7cfb6ac0 --To-tag as1df42b05 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117034@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK167dd99b Max-Forwards: 70 From: ;tag=as7cfb6ac0 To: ;tag=as1df42b05 Contact: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 10586 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117034@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39c211e6 Max-Forwards: 70 From: ;tag=as7cfb6ac0 To: Contact: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117034@178.62.121.41", nonce="225462f9", response="823804921a1668921d26633d148686a9" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637470277 637470278 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39c211e6;received=159.65.48.104 From: ;tag=as7cfb6ac0 To: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39c211e6;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7cfb6ac0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 (Checking To) --From tag as7cfb6ac0 --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13174] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13174] http.c: HTTP Request URI is /ari/channels/213007?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117033&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13174] http.c: match request [ari/channels/213007] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13174] http.c: match request [ari/channels/213007] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13174] http.c: match request [ari/channels/213007] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13174] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13174] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Finding handler for channels/213007 [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Finding handler for 213007 [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking channels create: Didn't match 213007 [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13174] res_ari.c: Checking channels externalMedia: Didn't match 213007 [Aug 18 10:33:50] DEBUG[13174] res_ari.c: No explicit handler found for 213007. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: Allocating new SIP dialog for 0d54090232075e1d6f617ed91cb15eb0@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13174] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c74010590' [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) RTP allocated port 14980 [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE creating session 0.0.0.0:14980 (14980) [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE create [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE add system candidates [Aug 18 10:33:50] DEBUG[13174] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13174] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE add candidate: 159.65.48.104:14980, 2130706431 [Aug 18 10:33:50] DEBUG[13174] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13174] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE add candidate: 10.131.0.10:14980, 2130706431 [Aug 18 10:33:50] DEBUG[13174] rtp_engine.c: RTP instance '0x7f0c74010590' is setup and ready to go [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) ICE stopped [Aug 18 10:33:50] DEBUG[13174] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13174] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13174] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13174] res_rtp_asterisk.c: (0x7f0c74010590) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13174] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13174] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: SIP call-id changed from '0d54090232075e1d6f617ed91cb15eb0@127.0.1.1:5060' to '398559732fb8625271bea90231b90490@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13174] stasis.c: Creating topic. name: channel:213007, detail: [Aug 18 10:33:50] DEBUG[13174] stasis.c: Topic 'channel:213007': 0x7f0c74016ed0 created [Aug 18 10:33:50] DEBUG[13174] stasis.c: Creating topic. name: cache:61/channel:213007, detail: [Aug 18 10:33:50] DEBUG[13174] stasis.c: Topic 'cache:61/channel:213007': 0x7f0c740170d0 created [Aug 18 10:33:50] DEBUG[13174] channel.c: Channel 0x7f0c74015470 'SIP/zvonobot-0000002a' allocated [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13174] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13174] res_stasis.c: calls_0: Subscribing to 213007 [Aug 18 10:33:50] DEBUG[13174] stasis/app.c: Channel '213007' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Outgoing Call for 79821117033 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13174] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13177] chan_sip.c: Audio is at 14980 [Aug 18 10:33:50] VERBOSE[13177] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13177] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13177] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13174] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Initializing initreq for method INVITE - callid 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117033@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62462f8d [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 3 [ 52]: From: ;tag=as686a751a [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 6 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13177] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117033@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62462f8d Max-Forwards: 70 From: ;tag=as686a751a To: Contact: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 555834509 555834509 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:50] DEBUG[13177] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13177] dial.c: Called zvonobot/79821117033 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62462f8d;received=159.65.48.104 From: ;tag=as686a751a To: ;tag=as2c6f6216 Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f11f39e" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62462f8d;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as686a751a [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2c6f6216 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f11f39e" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 (Checking To) --From tag as686a751a --To-tag as2c6f6216 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '398559732fb8625271bea90231b90490@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117033@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62462f8d Max-Forwards: 70 From: ;tag=as686a751a To: ;tag=as2c6f6216 Contact: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 14980 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117033@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4 Max-Forwards: 70 From: ;tag=as686a751a To: Contact: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117033@178.62.121.41", nonce="2f11f39e", response="b53e96c3e2db9316f59461e037b4521c" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 555834509 555834510 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 From: ;tag=as686a751a To: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as686a751a [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 (Checking To) --From tag as686a751a --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #10 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '398559732fb8625271bea90231b90490@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13179] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13179] http.c: HTTP Request URI is /ari/channels/213008?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117032&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13179] http.c: match request [ari/channels/213008] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13179] http.c: match request [ari/channels/213008] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13179] http.c: match request [ari/channels/213008] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13179] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13179] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Finding handler for channels/213008 [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Finding handler for 213008 [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking channels create: Didn't match 213008 [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13179] res_ari.c: Checking channels externalMedia: Didn't match 213008 [Aug 18 10:33:50] DEBUG[13179] res_ari.c: No explicit handler found for 213008. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: Allocating new SIP dialog for 2ab012f737d5f68750e7dd4e17f16100@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13179] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c020d90' [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP allocated port 15574 [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE creating session 0.0.0.0:15574 (15574) [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE create [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE add system candidates [Aug 18 10:33:50] DEBUG[13179] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13179] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE add candidate: 159.65.48.104:15574, 2130706431 [Aug 18 10:33:50] DEBUG[13179] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13179] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE add candidate: 10.131.0.10:15574, 2130706431 [Aug 18 10:33:50] DEBUG[13179] rtp_engine.c: RTP instance '0x7f0c7c020d90' is setup and ready to go [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE stopped [Aug 18 10:33:50] DEBUG[13179] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13179] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13179] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13179] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13179] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13179] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: SIP call-id changed from '2ab012f737d5f68750e7dd4e17f16100@127.0.1.1:5060' to '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13179] stasis.c: Creating topic. name: channel:213008, detail: [Aug 18 10:33:50] DEBUG[13179] stasis.c: Topic 'channel:213008': 0x7f0c7c016850 created [Aug 18 10:33:50] DEBUG[13179] stasis.c: Creating topic. name: cache:62/channel:213008, detail: [Aug 18 10:33:50] DEBUG[13179] stasis.c: Topic 'cache:62/channel:213008': 0x7f0c7c0259b0 created [Aug 18 10:33:50] DEBUG[13179] channel.c: Channel 0x7f0c7c0282f0 'SIP/zvonobot-0000002b' allocated [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13179] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13179] res_stasis.c: calls_0: Subscribing to 213008 [Aug 18 10:33:50] DEBUG[13179] stasis/app.c: Channel '213008' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Outgoing Call for 79821117032 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13179] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13183] chan_sip.c: Audio is at 15574 [Aug 18 10:33:50] VERBOSE[13183] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13183] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13183] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Initializing initreq for method INVITE - callid 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117032@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02acfa94 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 3 [ 52]: From: ;tag=as6be1179a [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 6 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13183] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117032@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02acfa94 Max-Forwards: 70 From: ;tag=as6be1179a To: Contact: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1720474686 1720474686 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15574 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:50] DEBUG[13183] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[13179] http.c: HTTP closing session. Top level [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02acfa94;received=159.65.48.104 From: ;tag=as6be1179a To: ;tag=as12a15c8c Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c4d0d10" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02acfa94;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as12a15c8c [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c4d0d10" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag as12a15c8c [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117032@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02acfa94 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as12a15c8c Contact: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 15574 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117032@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6979eb32 Max-Forwards: 70 From: ;tag=as6be1179a To: Contact: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41", nonce="7c4d0d10", response="daec26e2fcce26485b3f8901735c14bb" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1720474686 1720474687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15574 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13183] dial.c: Called zvonobot/79821117032 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6979eb32;received=159.65.48.104 From: ;tag=as6be1179a To: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[13184] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6979eb32;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13184] http.c: HTTP Request URI is /ari/channels/213005?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117035&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13184] http.c: match request [ari/channels/213005] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13184] http.c: match request [ari/channels/213005] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13184] http.c: match request [ari/channels/213005] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13184] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13184] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Finding handler for channels/213005 [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Finding handler for 213005 [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking channels create: Didn't match 213005 [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13184] res_ari.c: Checking channels externalMedia: Didn't match 213005 [Aug 18 10:33:50] DEBUG[13184] res_ari.c: No explicit handler found for 213005. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: Allocating new SIP dialog for 709b7ea3287d2dc53a0dcd42238977d0@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13184] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8403cbb0' [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) RTP allocated port 10930 [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE creating session 0.0.0.0:10930 (10930) [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE create [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE add system candidates [Aug 18 10:33:50] DEBUG[13184] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13184] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE add candidate: 159.65.48.104:10930, 2130706431 [Aug 18 10:33:50] DEBUG[13184] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13184] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE add candidate: 10.131.0.10:10930, 2130706431 [Aug 18 10:33:50] DEBUG[13184] rtp_engine.c: RTP instance '0x7f0c8403cbb0' is setup and ready to go [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) ICE stopped [Aug 18 10:33:50] DEBUG[13184] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13184] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13184] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13184] res_rtp_asterisk.c: (0x7f0c8403cbb0) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13184] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13184] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: SIP call-id changed from '709b7ea3287d2dc53a0dcd42238977d0@127.0.1.1:5060' to '0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13184] stasis.c: Creating topic. name: channel:213005, detail: [Aug 18 10:33:50] DEBUG[13184] stasis.c: Topic 'channel:213005': 0x7f0c84048590 created [Aug 18 10:33:50] DEBUG[13184] stasis.c: Creating topic. name: cache:63/channel:213005, detail: [Aug 18 10:33:50] DEBUG[13184] stasis.c: Topic 'cache:63/channel:213005': 0x7f0c84047450 created [Aug 18 10:33:50] DEBUG[13184] channel.c: Channel 0x7f0c84045ed0 'SIP/zvonobot-0000002c' allocated [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13184] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13184] res_stasis.c: calls_0: Subscribing to 213005 [Aug 18 10:33:50] DEBUG[13184] stasis/app.c: Channel '213005' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Outgoing Call for 79821117035 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13184] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13184] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13187] chan_sip.c: Audio is at 10930 [Aug 18 10:33:50] VERBOSE[13187] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13187] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13187] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Initializing initreq for method INVITE - callid 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117035@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ad28c9d [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 3 [ 52]: From: ;tag=as1b5137d9 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 6 [ 60]: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13187] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117035@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ad28c9d Max-Forwards: 70 From: ;tag=as1b5137d9 To: Contact: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1247977229 1247977229 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10930 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:50] DEBUG[13187] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ad28c9d;received=159.65.48.104 From: ;tag=as1b5137d9 To: ;tag=as3871b097 Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65f442e7" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ad28c9d;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1b5137d9 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3871b097 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65f442e7" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 (Checking To) --From tag as1b5137d9 --To-tag as3871b097 [Aug 18 10:33:50] VERBOSE[13187] dial.c: Called zvonobot/79821117035 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117035@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ad28c9d Max-Forwards: 70 From: ;tag=as1b5137d9 To: ;tag=as3871b097 Contact: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 10930 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117035@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39473786 Max-Forwards: 70 From: ;tag=as1b5137d9 To: Contact: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117035@178.62.121.41", nonce="65f442e7", response="7062ed749db471924b38e43497b81c30" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1247977229 1247977230 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10930 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39473786;received=159.65.48.104 From: ;tag=as1b5137d9 To: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39473786;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1b5137d9 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060 (Checking To) --From tag as1b5137d9 --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #2 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0d58c1e7283e425c4d2ff5534dda8681@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13190] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13190] http.c: HTTP Request URI is /ari/channels/213010?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117030&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13190] http.c: match request [ari/channels/213010] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13190] http.c: match request [ari/channels/213010] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13190] http.c: match request [ari/channels/213010] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13190] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13190] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Finding handler for channels/213010 [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Finding handler for 213010 [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking channels create: Didn't match 213010 [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13190] res_ari.c: Checking channels externalMedia: Didn't match 213010 [Aug 18 10:33:50] DEBUG[13190] res_ari.c: No explicit handler found for 213010. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: Allocating new SIP dialog for 6c3a9d620a7a419e5fa7f9492330fcb1@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13190] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c020490' [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) RTP allocated port 19548 [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE creating session 0.0.0.0:19548 (19548) [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE create [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE add system candidates [Aug 18 10:33:50] DEBUG[13190] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13190] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE add candidate: 159.65.48.104:19548, 2130706431 [Aug 18 10:33:50] DEBUG[13190] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13190] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE add candidate: 10.131.0.10:19548, 2130706431 [Aug 18 10:33:50] DEBUG[13190] rtp_engine.c: RTP instance '0x7f0c8c020490' is setup and ready to go [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) ICE stopped [Aug 18 10:33:50] DEBUG[13190] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13190] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13190] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13190] res_rtp_asterisk.c: (0x7f0c8c020490) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13190] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13190] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: SIP call-id changed from '6c3a9d620a7a419e5fa7f9492330fcb1@127.0.1.1:5060' to '56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13190] stasis.c: Creating topic. name: channel:213010, detail: [Aug 18 10:33:50] DEBUG[13190] stasis.c: Topic 'channel:213010': 0x7f0c8c01f8d0 created [Aug 18 10:33:50] DEBUG[13190] stasis.c: Creating topic. name: cache:64/channel:213010, detail: [Aug 18 10:33:50] DEBUG[13190] stasis.c: Topic 'cache:64/channel:213010': 0x7f0c8c029300 created [Aug 18 10:33:50] DEBUG[13190] channel.c: Channel 0x7f0c8c027e40 'SIP/zvonobot-0000002d' allocated [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13190] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13190] res_stasis.c: calls_0: Subscribing to 213010 [Aug 18 10:33:50] DEBUG[13190] stasis/app.c: Channel '213010' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13190] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Outgoing Call for 79821117030 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13192] chan_sip.c: Audio is at 19548 [Aug 18 10:33:50] VERBOSE[13192] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13192] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13192] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13190] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Initializing initreq for method INVITE - callid 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117030@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06d1dd31 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 3 [ 52]: From: ;tag=as54e004b1 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 6 [ 60]: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13192] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117030@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06d1dd31 Max-Forwards: 70 From: ;tag=as54e004b1 To: Contact: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 953853099 953853099 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Aug 18 10:33:50] DEBUG[13192] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13192] dial.c: Called zvonobot/79821117030 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06d1dd31;received=159.65.48.104 From: ;tag=as54e004b1 To: ;tag=as471ac8a9 Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b6aa3cb" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06d1dd31;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54e004b1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as471ac8a9 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b6aa3cb" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 (Checking To) --From tag as54e004b1 --To-tag as471ac8a9 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117030@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06d1dd31 Max-Forwards: 70 From: ;tag=as54e004b1 To: ;tag=as471ac8a9 Contact: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 19548 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117030@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8 Max-Forwards: 70 From: ;tag=as54e004b1 To: Contact: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117030@178.62.121.41", nonce="3b6aa3cb", response="d0b7427e54c19fdc94523f008944289d" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 953853099 953853100 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 From: ;tag=as54e004b1 To: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54e004b1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 (Checking To) --From tag as54e004b1 --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13194] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13194] http.c: HTTP Request URI is /ari/channels/213012?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117028&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13194] http.c: match request [ari/channels/213012] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13194] http.c: match request [ari/channels/213012] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13194] http.c: match request [ari/channels/213012] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13194] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13194] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Finding handler for channels/213012 [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Finding handler for 213012 [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking channels create: Didn't match 213012 [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13194] res_ari.c: Checking channels externalMedia: Didn't match 213012 [Aug 18 10:33:50] DEBUG[13194] res_ari.c: No explicit handler found for 213012. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: Allocating new SIP dialog for 792fa54c083cbd5c087442510ea28971@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13194] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c94022610' [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) RTP allocated port 14220 [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE creating session 0.0.0.0:14220 (14220) [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE create [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE add system candidates [Aug 18 10:33:50] DEBUG[13194] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13194] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE add candidate: 159.65.48.104:14220, 2130706431 [Aug 18 10:33:50] DEBUG[13194] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13194] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE add candidate: 10.131.0.10:14220, 2130706431 [Aug 18 10:33:50] DEBUG[13194] rtp_engine.c: RTP instance '0x7f0c94022610' is setup and ready to go [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) ICE stopped [Aug 18 10:33:50] DEBUG[13194] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13194] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13194] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13194] res_rtp_asterisk.c: (0x7f0c94022610) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13194] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13194] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: SIP call-id changed from '792fa54c083cbd5c087442510ea28971@127.0.1.1:5060' to '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13194] stasis.c: Creating topic. name: channel:213012, detail: [Aug 18 10:33:50] DEBUG[13194] stasis.c: Topic 'channel:213012': 0x7f0c9402ae80 created [Aug 18 10:33:50] DEBUG[13194] stasis.c: Creating topic. name: cache:65/channel:213012, detail: [Aug 18 10:33:50] DEBUG[13194] stasis.c: Topic 'cache:65/channel:213012': 0x7f0c940294e0 created [Aug 18 10:33:50] DEBUG[13194] channel.c: Channel 0x7f0c94027db0 'SIP/zvonobot-0000002e' allocated [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13194] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13194] res_stasis.c: calls_0: Subscribing to 213012 [Aug 18 10:33:50] DEBUG[13194] stasis/app.c: Channel '213012' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Outgoing Call for 79821117028 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13194] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13195] chan_sip.c: Audio is at 14220 [Aug 18 10:33:50] VERBOSE[13195] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13195] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13195] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Initializing initreq for method INVITE - callid 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117028@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c64de4b [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13194] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 3 [ 52]: From: ;tag=as510b84fe [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 6 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13195] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117028@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c64de4b Max-Forwards: 70 From: ;tag=as510b84fe To: Contact: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1638258584 1638258584 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14220 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:50] DEBUG[13195] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13195] dial.c: Called zvonobot/79821117028 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c64de4b;received=159.65.48.104 From: ;tag=as510b84fe To: ;tag=as5dd3a7f5 Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="589c1c16" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c64de4b;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as510b84fe [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5dd3a7f5 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="589c1c16" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 (Checking To) --From tag as510b84fe --To-tag as5dd3a7f5 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117028@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c64de4b Max-Forwards: 70 From: ;tag=as510b84fe To: ;tag=as5dd3a7f5 Contact: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 14220 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117028@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK12ed07ac Max-Forwards: 70 From: ;tag=as510b84fe To: Contact: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117028@178.62.121.41", nonce="589c1c16", response="3a6363b41ad25fdbb6589c43c42b58c4" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1638258584 1638258585 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14220 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK12ed07ac;received=159.65.48.104 From: ;tag=as510b84fe To: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK12ed07ac;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as510b84fe [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 (Checking To) --From tag as510b84fe --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13197] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13197] http.c: HTTP Request URI is /ari/channels/213009?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117031&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13197] http.c: match request [ari/channels/213009] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13197] http.c: match request [ari/channels/213009] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13197] http.c: match request [ari/channels/213009] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13197] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13197] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Finding handler for channels/213009 [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Finding handler for 213009 [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking channels create: Didn't match 213009 [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13197] res_ari.c: Checking channels externalMedia: Didn't match 213009 [Aug 18 10:33:50] DEBUG[13197] res_ari.c: No explicit handler found for 213009. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: Allocating new SIP dialog for 12412a460c6a653f030295427b6e92c0@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13197] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca800cb50' [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP allocated port 17902 [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE creating session 0.0.0.0:17902 (17902) [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE create [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE add system candidates [Aug 18 10:33:50] DEBUG[13197] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13197] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE add candidate: 159.65.48.104:17902, 2130706431 [Aug 18 10:33:50] DEBUG[13197] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13197] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE add candidate: 10.131.0.10:17902, 2130706431 [Aug 18 10:33:50] DEBUG[13197] rtp_engine.c: RTP instance '0x7f0ca800cb50' is setup and ready to go [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE stopped [Aug 18 10:33:50] DEBUG[13197] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13197] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13197] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13197] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13197] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13197] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: SIP call-id changed from '12412a460c6a653f030295427b6e92c0@127.0.1.1:5060' to '79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13197] stasis.c: Creating topic. name: channel:213009, detail: [Aug 18 10:33:50] DEBUG[13197] stasis.c: Topic 'channel:213009': 0x7f0ca8026df0 created [Aug 18 10:33:50] DEBUG[13197] stasis.c: Creating topic. name: cache:66/channel:213009, detail: [Aug 18 10:33:50] DEBUG[13197] stasis.c: Topic 'cache:66/channel:213009': 0x7f0ca8027040 created [Aug 18 10:33:50] DEBUG[13197] channel.c: Channel 0x7f0ca8024790 'SIP/zvonobot-0000002f' allocated [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13197] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13197] res_stasis.c: calls_0: Subscribing to 213009 [Aug 18 10:33:50] DEBUG[13197] stasis/app.c: Channel '213009' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Outgoing Call for 79821117031 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13197] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13198] chan_sip.c: Audio is at 17902 [Aug 18 10:33:50] VERBOSE[13198] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13198] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13198] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Initializing initreq for method INVITE - callid 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117031@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ef2f350 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 3 [ 52]: From: ;tag=as73737e94 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 6 [ 60]: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13197] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13198] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117031@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ef2f350 Max-Forwards: 70 From: ;tag=as73737e94 To: Contact: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 621248212 621248212 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:50] DEBUG[13198] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13198] dial.c: Called zvonobot/79821117031 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ef2f350;received=159.65.48.104 From: ;tag=as73737e94 To: ;tag=as73e79746 Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b9a7b89" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ef2f350;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as73737e94 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as73e79746 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b9a7b89" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 (Checking To) --From tag as73737e94 --To-tag as73e79746 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117031@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ef2f350 Max-Forwards: 70 From: ;tag=as73737e94 To: ;tag=as73e79746 Contact: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 17902 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117031@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa Max-Forwards: 70 From: ;tag=as73737e94 To: Contact: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117031@178.62.121.41", nonce="6b9a7b89", response="902b980065f093be7c1be72aa9a065d7" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 621248212 621248213 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 From: ;tag=as73737e94 To: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as73737e94 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 (Checking To) --From tag as73737e94 --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #10 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13200] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13200] http.c: HTTP Request URI is /ari/channels/213011?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117029&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13200] http.c: match request [ari/channels/213011] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13200] http.c: match request [ari/channels/213011] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13200] http.c: match request [ari/channels/213011] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13200] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13200] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Finding handler for channels/213011 [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Finding handler for 213011 [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking channels create: Didn't match 213011 [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13200] res_ari.c: Checking channels externalMedia: Didn't match 213011 [Aug 18 10:33:50] DEBUG[13200] res_ari.c: No explicit handler found for 213011. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: Allocating new SIP dialog for 13352371653b8220090490bc403b4f79@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13200] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9801aec0' [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP allocated port 13092 [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE creating session 0.0.0.0:13092 (13092) [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE create [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE add system candidates [Aug 18 10:33:50] DEBUG[13200] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13200] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE add candidate: 159.65.48.104:13092, 2130706431 [Aug 18 10:33:50] DEBUG[13200] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13200] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE add candidate: 10.131.0.10:13092, 2130706431 [Aug 18 10:33:50] DEBUG[13200] rtp_engine.c: RTP instance '0x7f0c9801aec0' is setup and ready to go [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE stopped [Aug 18 10:33:50] DEBUG[13200] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13200] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13200] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13200] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP setup on RTP instance [Aug 18 10:33:50] VERBOSE[13200] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13200] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: SIP call-id changed from '13352371653b8220090490bc403b4f79@127.0.1.1:5060' to '710394295318048c14806fba23b501f2@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13200] stasis.c: Creating topic. name: channel:213011, detail: [Aug 18 10:33:50] DEBUG[13200] stasis.c: Topic 'channel:213011': 0x7f0c98025620 created [Aug 18 10:33:50] DEBUG[13200] stasis.c: Creating topic. name: cache:67/channel:213011, detail: [Aug 18 10:33:50] DEBUG[13200] stasis.c: Topic 'cache:67/channel:213011': 0x7f0c980241b0 created [Aug 18 10:33:50] DEBUG[13200] channel.c: Channel 0x7f0c980222e0 'SIP/zvonobot-00000030' allocated [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13200] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13200] res_stasis.c: calls_0: Subscribing to 213011 [Aug 18 10:33:50] DEBUG[13200] stasis/app.c: Channel '213011' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Outgoing Call for 79821117029 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13200] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13201] chan_sip.c: Audio is at 13092 [Aug 18 10:33:50] VERBOSE[13201] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13201] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13201] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Initializing initreq for method INVITE - callid 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117029@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3fb0d987 [Aug 18 10:33:50] DEBUG[13200] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 3 [ 52]: From: ;tag=as08bf07d1 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 6 [ 60]: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13201] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117029@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3fb0d987 Max-Forwards: 70 From: ;tag=as08bf07d1 To: Contact: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 572235632 572235632 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13092 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:33:50] DEBUG[13201] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[13201] dial.c: Called zvonobot/79821117029 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3fb0d987;received=159.65.48.104 From: ;tag=as08bf07d1 To: ;tag=as56121eda Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ef41625" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3fb0d987;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08bf07d1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56121eda [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ef41625" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 (Checking To) --From tag as08bf07d1 --To-tag as56121eda [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '710394295318048c14806fba23b501f2@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117029@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3fb0d987 Max-Forwards: 70 From: ;tag=as08bf07d1 To: ;tag=as56121eda Contact: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 13092 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117029@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK417bdd1e Max-Forwards: 70 From: ;tag=as08bf07d1 To: Contact: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117029@178.62.121.41", nonce="2ef41625", response="3cca894767e459c5b993e75b58a2249c" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 572235632 572235633 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13092 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK417bdd1e;received=159.65.48.104 From: ;tag=as08bf07d1 To: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK417bdd1e;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08bf07d1 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 (Checking To) --From tag as08bf07d1 --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '710394295318048c14806fba23b501f2@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[13203] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13203] http.c: HTTP Request URI is /ari/channels/213013?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117027&callerId=74950493843 [Aug 18 10:33:50] DEBUG[13203] http.c: match request [ari/channels/213013] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13203] http.c: match request [ari/channels/213013] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13203] http.c: match request [ari/channels/213013] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13203] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13203] http.c: HTTP consuming request body [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Finding handler for channels/213013 [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Finding handler for 213013 [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking channels create: Didn't match 213013 [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13203] res_ari.c: Checking channels externalMedia: Didn't match 213013 [Aug 18 10:33:50] DEBUG[13203] res_ari.c: No explicit handler found for 213013. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Allocating new SIP dialog for 71deed180cbf11284e561edd6f0a32c3@127.0.1.1:5060 - OPTIONS (No RTP) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP call-id changed from '71deed180cbf11284e561edd6f0a32c3@127.0.1.1:5060' to '56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Initializing initreq for method OPTIONS - callid 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 33]: OPTIONS sip:178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4d4cb804 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as7e00d300 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 23]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 42]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 60]: Call-ID: 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: OPTIONS sip:178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4d4cb804 Max-Forwards: 70 From: "asterisk" ;tag=as7e00d300 To: Contact: Call-ID: 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49f889c9;received=159.65.48.104 From: ;tag=as67678dc7 To: ;tag=as31a963bc Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 389747437 389747437 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10788 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49f889c9;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as67678dc7 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31a963bc [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 389747437 389747437 IN IP4 178.62.121.41 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10788 RTP/AVP 0 8 101 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking To) --From tag as67678dc7 --To-tag as31a963bc [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Got SDP version 389747437 and unique parts [root 389747437 IN IP4 178.62.121.41] [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 389747437 389747437 IN IP4 178.62.121.41... OK. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE set role failed; no ice instance [Aug 18 10:33:50] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP setting address on RTP instance [Aug 18 10:33:50] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c100223b0 -- Strict RTP learning after remote address set to: 178.62.121.41:10788 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10788 [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb000e8a8) from 0x7f0c147e2330 to 0x7f0c1007bf18 [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00795f8) from 0x7f0c147e2330 to 0x7f0c1007bf18 [Aug 18 10:33:50] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0076b08) from 0x7f0c147e2330 to 0x7f0c1007bf18 [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP ignoring duplicate property [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:50] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000012 setting read format path: alaw -> alaw [Aug 18 10:33:50] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000012 setting write format path: alaw -> alaw [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS - ast_rtp_activate rtp=0x7f0c100223b0 - setup and perform DTLS' [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100223b0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:50] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100223b0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Strict routing enforced for session 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:50] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:50] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117057@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2688dd50 Max-Forwards: 70 From: ;tag=as67678dc7 To: ;tag=as31a963bc Contact: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Session timer started: 15 - 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 1768000ms [Aug 18 10:33:50] VERBOSE[12968] dial.c: SIP/zvonobot-00000012 answered [Aug 18 10:33:50] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:50] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:50] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:50] VERBOSE[12968] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000012 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4d4cb804;received=159.65.48.104 From: "asterisk" ;tag=as7e00d300 To: ;tag=as2c12ccd6 Call-ID: 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4d4cb804;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as7e00d300 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 38]: To: ;tag=as2c12ccd6 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 (Checking To) --From tag as7e00d300 --To-tag as2c12ccd6 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[12968] stasis/app.c: Channel '212983' is 2 interested in calls_0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Destroying SIP dialog 56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '56e3f4335077fe5d38154ae13c75055c@159.65.48.104:5060' Method: OPTIONS [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: Allocating new SIP dialog for 7bf9161917a8540a0c3bca356e3acdeb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:50] DEBUG[13203] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca402d940' [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) RTP allocated port 14404 [Aug 18 10:33:50] DEBUG[13204] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13204] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE creating session 0.0.0.0:14404 (14404) [Aug 18 10:33:50] DEBUG[13204] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE create [Aug 18 10:33:50] DEBUG[13204] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13204] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13204] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13204] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] DEBUG[13204] stasis.c: Creating topic. name: bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6, detail: [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE add system candidates [Aug 18 10:33:50] DEBUG[13204] stasis.c: Topic 'bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6': 0x7f0cb0036d50 created [Aug 18 10:33:50] DEBUG[13203] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13204] stasis.c: Creating topic. name: cache:68/bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6, detail: [Aug 18 10:33:50] DEBUG[13203] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE add candidate: 159.65.48.104:14404, 2130706431 [Aug 18 10:33:50] DEBUG[13203] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13203] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13204] stasis.c: Topic 'cache:68/bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6': 0x7f0cb006aa70 created [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE add candidate: 10.131.0.10:14404, 2130706431 [Aug 18 10:33:50] DEBUG[13203] rtp_engine.c: RTP instance '0x7f0ca402d940' is setup and ready to go [Aug 18 10:33:50] DEBUG[13204] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as two channels are required [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) ICE stopped [Aug 18 10:33:50] DEBUG[13204] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13203] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:50] DEBUG[13204] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13203] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:50] DEBUG[13204] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:50] DEBUG[13204] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13203] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:50] DEBUG[13204] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology constructor [Aug 18 10:33:50] DEBUG[13203] res_rtp_asterisk.c: (0x7f0ca402d940) RTCP setup on RTP instance [Aug 18 10:33:50] DEBUG[13204] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology start [Aug 18 10:33:50] VERBOSE[13203] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:50] DEBUG[13203] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:50] DEBUG[13204] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13204] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13205] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:50] DEBUG[13205] http.c: HTTP Request URI is /ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/addChannel?channel=212983 [Aug 18 10:33:50] DEBUG[13205] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13205] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13205] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/addChannel] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: SIP call-id changed from '7bf9161917a8540a0c3bca356e3acdeb@127.0.1.1:5060' to '50b3abe5238219eb79521a71216dea65@159.65.48.104:5060' [Aug 18 10:33:50] DEBUG[13205] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Finding handler for bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/addChannel [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13203] stasis.c: Creating topic. name: channel:213013, detail: [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13203] stasis.c: Topic 'channel:213013': 0x7f0ca4024600 created [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13203] stasis.c: Creating topic. name: cache:69/channel:213013, detail: [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Finding handler for 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13203] stasis.c: Topic 'cache:69/channel:213013': 0x7f0ca4035020 created [Aug 18 10:33:50] DEBUG[13205] res_ari.c: No explicit handler found for 378d72c1-dd9d-472b-9f36-5c575a6102e6. Using wildcard bridgeId. [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Finding handler for addChannel [Aug 18 10:33:50] DEBUG[13205] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:50] DEBUG[13205] stasis/control.c: 212983: Sending channel add_to_bridge command [Aug 18 10:33:50] DEBUG[13203] channel.c: Channel 0x7f0ca4033100 'SIP/zvonobot-00000031' allocated [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:50] DEBUG[13203] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:50] DEBUG[13203] res_stasis.c: calls_0: Subscribing to 213013 [Aug 18 10:33:50] DEBUG[13203] stasis/app.c: Channel '213013' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13203] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13203] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Outgoing Call for 79821117027 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[13206] chan_sip.c: Audio is at 14404 [Aug 18 10:33:50] VERBOSE[13206] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[13206] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[13206] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Initializing initreq for method INVITE - callid 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117027@178.62.121.41 SIP/2.0 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ed0df05 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 3 [ 52]: From: ;tag=as1d2c553b [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 6 [ 60]: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:50 GMT [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:50] VERBOSE[13206] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117027@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ed0df05 Max-Forwards: 70 From: ;tag=as1d2c553b To: Contact: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1478400641 1478400641 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14404 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:33:50] DEBUG[13206] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ed0df05;received=159.65.48.104 From: ;tag=as1d2c553b To: ;tag=as1ee9c6dd Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09f7ce32" Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ed0df05;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1d2c553b [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1ee9c6dd [Aug 18 10:33:50] VERBOSE[13206] dial.c: Called zvonobot/79821117027 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09f7ce32" [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 (Checking To) --From tag as1d2c553b --To-tag as1ee9c6dd [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Stopping retransmission on '50b3abe5238219eb79521a71216dea65@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117027@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4ed0df05 Max-Forwards: 70 From: ;tag=as1d2c553b To: ;tag=as1ee9c6dd Contact: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Audio is at 14404 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117027@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35a9e8c5 Max-Forwards: 70 From: ;tag=as1d2c553b To: Contact: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117027@178.62.121.41", nonce="09f7ce32", response="90f025a10fe8cd10cba15fb8807384f2" Date: Wed, 18 Aug 2021 10:33:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1478400641 1478400642 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14404 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35a9e8c5;received=159.65.48.104 From: ;tag=as1d2c553b To: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35a9e8c5;received=159.65.48.104 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1d2c553b [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:50] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50b3abe5238219eb79521a71216dea65@159.65.48.104:5060 (Checking To) --From tag as1d2c553b --To-tag [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '50b3abe5238219eb79521a71216dea65@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:50] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:50] DEBUG[12968] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000012 [Aug 18 10:33:50] DEBUG[12968] stasis/control.c: 212983: Adding to bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:33:50] DEBUG[12968] stasis/app.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13208] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is joining [Aug 18 10:33:50] DEBUG[13208] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pushing 0x7f0c080248a0(SIP/zvonobot-00000012) [Aug 18 10:33:50] VERBOSE[13208] bridge_channel.c: Channel SIP/zvonobot-00000012 joined 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:33:50] DEBUG[13208] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as two channels are required [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 is already using the new technology. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is joining simple_bridge technology [Aug 18 10:33:50] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP changing ssrc from 1755855706 to 1346315696 due to a source change [Aug 18 10:33:50] DEBUG[12968] stasis/app.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' is 2 interested in calls_0 [Aug 18 10:33:50] DEBUG[13205] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:50] DEBUG[13205] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13209] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13209] http.c: HTTP Request URI is /ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/record?name=212983_sHOlyejrGUkfTvaheXVVENZBSbbPfMiE&format=wav [Aug 18 10:33:50] DEBUG[13209] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/record] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13209] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/record] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13209] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/record] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13209] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Finding handler for bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Finding handler for 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13209] res_ari.c: No explicit handler found for 378d72c1-dd9d-472b-9f36-5c575a6102e6. Using wildcard bridgeId. [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Finding handler for record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:50] DEBUG[13209] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:50] DEBUG[13209] stasis.c: Creating topic. name: channel:1629282830.60, detail: [Aug 18 10:33:50] DEBUG[13209] stasis.c: Topic 'channel:1629282830.60': 0x7f0c1000def0 created [Aug 18 10:33:50] DEBUG[13209] stasis.c: Creating topic. name: cache:70/channel:1629282830.60, detail: [Aug 18 10:33:50] DEBUG[13209] stasis.c: Topic 'cache:70/channel:1629282830.60': 0x7f0c1002d480 created [Aug 18 10:33:50] DEBUG[13209] channel.c: Channel 0x7f0c1002e710 'Recorder/ARI-00000003;1' allocated [Aug 18 10:33:50] DEBUG[13209] stasis.c: Creating topic. name: channel:1629282830.61, detail: [Aug 18 10:33:50] DEBUG[13209] stasis.c: Topic 'channel:1629282830.61': 0x7f0c10030630 created [Aug 18 10:33:50] DEBUG[13209] stasis.c: Creating topic. name: cache:71/channel:1629282830.61, detail: [Aug 18 10:33:50] DEBUG[13209] stasis.c: Topic 'cache:71/channel:1629282830.61': 0x7f0c10037770 created [Aug 18 10:33:50] DEBUG[13209] channel.c: Channel 0x7f0c10035e20 'Recorder/ARI-00000003;2' allocated [Aug 18 10:33:50] DEBUG[13209] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:50] DEBUG[13210] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining [Aug 18 10:33:50] DEBUG[13210] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pushing 0x7f0c100369e0(Recorder/ARI-00000003;2) [Aug 18 10:33:50] DEBUG[13210] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:50] VERBOSE[13210] bridge_channel.c: Channel Recorder/ARI-00000003;2 joined 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:33:50] DEBUG[13210] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6'. Checking compatability for channels 'SIP/zvonobot-00000012' and 'Recorder/ARI-00000003;2' [Aug 18 10:33:50] DEBUG[13210] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as could not get details [Aug 18 10:33:50] DEBUG[13210] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13210] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13210] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13210] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13210] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 is already using the new technology. [Aug 18 10:33:50] DEBUG[13210] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining simple_bridge technology [Aug 18 10:33:50] DEBUG[13210] channel.c: Channel Recorder/ARI-00000003;2 setting read format path: slin -> slin [Aug 18 10:33:50] DEBUG[13210] channel.c: Channel SIP/zvonobot-00000012 setting write format path: slin -> alaw [Aug 18 10:33:50] DEBUG[13210] channel.c: Channel SIP/zvonobot-00000012 setting read format path: alaw -> slin [Aug 18 10:33:50] DEBUG[13210] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:33:50] DEBUG[13209] res_stasis_recording.c: 1629282830.60: Sending record(212983_sHOlyejrGUkfTvaheXVVENZBSbbPfMiE.wav) command [Aug 18 10:33:50] DEBUG[13212] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13211] app.c: play_and_record: , /var/spool/asterisk/recording/212983_sHOlyejrGUkfTvaheXVVENZBSbbPfMiE, 'wav' [Aug 18 10:33:50] DEBUG[13212] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:50] DEBUG[13211] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:50] DEBUG[13212] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13212] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13212] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13212] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13209] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13209] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13212] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] VERBOSE[13211] app.c: x=0, open writing: /var/spool/asterisk/recording/212983_sHOlyejrGUkfTvaheXVVENZBSbbPfMiE format: wav, 0x7f0c1c0313a0 [Aug 18 10:33:50] DEBUG[13212] stasis.c: Creating topic. name: bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee, detail: [Aug 18 10:33:50] DEBUG[13212] stasis.c: Topic 'bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee': 0x7f0c1800caa0 created [Aug 18 10:33:50] DEBUG[13212] stasis.c: Creating topic. name: cache:72/bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee, detail: [Aug 18 10:33:50] DEBUG[13212] stasis.c: Topic 'cache:72/bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee': 0x7f0c18012b90 created [Aug 18 10:33:50] DEBUG[13212] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' can not use native RTP bridge as two channels are required [Aug 18 10:33:50] DEBUG[13212] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13212] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13212] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:50] DEBUG[13212] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13212] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: calling simple_bridge technology constructor [Aug 18 10:33:50] DEBUG[13212] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: calling simple_bridge technology start [Aug 18 10:33:50] DEBUG[13212] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13212] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13213] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13213] http.c: HTTP Request URI is /ari/channels/212983/snoop?app=calls_0&spy=in [Aug 18 10:33:50] DEBUG[13213] http.c: match request [ari/channels/212983/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13213] http.c: match request [ari/channels/212983/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13213] http.c: match request [ari/channels/212983/snoop] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13213] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Finding handler for channels/212983/snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Finding handler for 212983 [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channels create: Didn't match 212983 [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channels externalMedia: Didn't match 212983 [Aug 18 10:33:50] DEBUG[13213] res_ari.c: No explicit handler found for 212983. Using wildcard channelId. [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Finding handler for snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:50] DEBUG[13213] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:50] DEBUG[13213] stasis.c: Creating topic. name: channel:1629282830.62, detail: [Aug 18 10:33:50] DEBUG[13213] stasis.c: Topic 'channel:1629282830.62': 0x7f0c240089d0 created [Aug 18 10:33:50] DEBUG[13213] stasis.c: Creating topic. name: cache:73/channel:1629282830.62, detail: [Aug 18 10:33:50] DEBUG[13213] stasis.c: Topic 'cache:73/channel:1629282830.62': 0x7f0c24007af0 created [Aug 18 10:33:50] DEBUG[13213] channel.c: Channel 0x7f0c2402e210 'Snoop/212983-00000001' allocated [Aug 18 10:33:50] DEBUG[13208] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6'. Checking compatability for channels 'SIP/zvonobot-00000012' and 'Recorder/ARI-00000003;2' [Aug 18 10:33:50] DEBUG[13208] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as channel 'SIP/zvonobot-00000012' has features which prevent it [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13208] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 is already using the new technology. [Aug 18 10:33:50] DEBUG[13213] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13213] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13217] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13217] http.c: HTTP Request URI is /ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play?media=sound%3Asilence%2F2 [Aug 18 10:33:50] DEBUG[13214] stasis/app.c: Channel '1629282830.62' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13217] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13217] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13217] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13217] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Finding handler for bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Finding handler for 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13217] res_ari.c: No explicit handler found for 378d72c1-dd9d-472b-9f36-5c575a6102e6. Using wildcard bridgeId. [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Finding handler for play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:50] DEBUG[13217] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:50] DEBUG[13217] stasis.c: Creating topic. name: channel:1629282830.63, detail: [Aug 18 10:33:50] DEBUG[13220] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13220] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212983&app=calls_0&format=slin16&external_host=127.0.0.1%3A50397 [Aug 18 10:33:50] DEBUG[13220] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13220] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13220] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13220] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13217] stasis.c: Topic 'channel:1629282830.63': 0x7f0c2c01b1b0 created [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 457 [Aug 18 10:33:50] DEBUG[13217] stasis.c: Creating topic. name: cache:74/channel:1629282830.63, detail: [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Finding handler for channels [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 457 [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13220] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:50] DEBUG[13220] netsock2.c: Splitting '127.0.0.1:50397' into... [Aug 18 10:33:50] DEBUG[13220] netsock2.c: ...host '127.0.0.1' and port '50397'. [Aug 18 10:33:50] DEBUG[13220] netsock2.c: Splitting '127.0.0.1:50397' into... [Aug 18 10:33:50] DEBUG[13220] netsock2.c: ...host '127.0.0.1' and port '50397'. [Aug 18 10:33:50] DEBUG[13220] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:50] DEBUG[13220] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c28011240' [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) RTP allocated port 12574 [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) ICE creating session 127.0.0.1:12574 (12574) [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) ICE create [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) ICE add system candidates [Aug 18 10:33:50] DEBUG[13220] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:50] DEBUG[13220] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) ICE add candidate: 159.65.48.104:12574, 2130706431 [Aug 18 10:33:50] DEBUG[13220] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:50] DEBUG[13220] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:50] DEBUG[13220] res_rtp_asterisk.c: (0x7f0c28011240) ICE add candidate: 10.131.0.10:12574, 2130706431 [Aug 18 10:33:50] DEBUG[13220] rtp_engine.c: RTP instance '0x7f0c28011240' is setup and ready to go [Aug 18 10:33:50] DEBUG[13220] stasis.c: Creating topic. name: channel:robot_212983, detail: [Aug 18 10:33:50] DEBUG[13217] stasis.c: Topic 'cache:74/channel:1629282830.63': 0x7f0c2c077310 created [Aug 18 10:33:50] DEBUG[13220] stasis.c: Topic 'channel:robot_212983': 0x7f0c2800a760 created [Aug 18 10:33:50] DEBUG[13220] stasis.c: Creating topic. name: cache:75/channel:robot_212983, detail: [Aug 18 10:33:50] DEBUG[13220] stasis.c: Topic 'cache:75/channel:robot_212983': 0x7f0c2807d180 created [Aug 18 10:33:50] DEBUG[13217] channel.c: Channel 0x7f0c2c00b5e0 'Announcer/ARI-00000004;1' allocated [Aug 18 10:33:50] DEBUG[13217] stasis.c: Creating topic. name: channel:1629282830.65, detail: [Aug 18 10:33:50] DEBUG[13217] stasis.c: Topic 'channel:1629282830.65': 0x7f0c2c00f460 created [Aug 18 10:33:50] DEBUG[13217] stasis.c: Creating topic. name: cache:76/channel:1629282830.65, detail: [Aug 18 10:33:50] DEBUG[13217] stasis.c: Topic 'cache:76/channel:1629282830.65': 0x7f0c2c00fcc0 created [Aug 18 10:33:50] DEBUG[13220] channel.c: Channel 0x7f0c2807fb90 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240' allocated [Aug 18 10:33:50] DEBUG[13220] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:50] VERBOSE[13220] res_rtp_asterisk.c: 0x7f0c28012d60 -- Strict RTP learning after remote address set to: 127.0.0.1:50397 [Aug 18 10:33:50] DEBUG[13217] channel.c: Channel 0x7f0c2c04ba30 'Announcer/ARI-00000004;2' allocated [Aug 18 10:33:50] DEBUG[13220] res_stasis.c: calls_0: Subscribing to robot_212983 [Aug 18 10:33:50] DEBUG[13217] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:50] DEBUG[13220] stasis/app.c: Channel 'robot_212983' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13217] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000004;1' [Aug 18 10:33:50] DEBUG[13220] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:50] DEBUG[13222] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2c00ad50(Announcer/ARI-00000004;2) is joining [Aug 18 10:33:50] VERBOSE[13221] dial.c: Called 127.0.0.1:50397 [Aug 18 10:33:50] DEBUG[13222] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pushing 0x7f0c2c00ad50(Announcer/ARI-00000004;2) [Aug 18 10:33:50] DEBUG[13220] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13222] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:50] VERBOSE[13222] bridge_channel.c: Channel Announcer/ARI-00000004;2 joined 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13222] bridge.c: Chose bridge technology softmix [Aug 18 10:33:50] VERBOSE[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: switching from simple_bridge technology to softmix [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology constructor [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c080248a0(SIP/zvonobot-00000012) to dummy bridge temporarily [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c100369e0(Recorder/ARI-00000003;2) to dummy bridge temporarily [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is leaving simple_bridge technology (dummy) [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology stop [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2c00ad50(Announcer/ARI-00000004;2) is joining softmix technology [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Announcer/ARI-00000004;2: [Aug 18 10:33:50] DEBUG[13222] channel.c: Channel Announcer/ARI-00000004;2 setting write format path: slin -> slin [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Announcer/ARI-00000004;2: [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Announcer/ARI-00000004;2: Not in SFU mode [Aug 18 10:33:50] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50397 [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is joining softmix technology [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: SIP/zvonobot-00000012: [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: SIP/zvonobot-00000012: [Aug 18 10:33:50] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50397 - state 2 (In use) [Aug 18 10:33:50] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50397, detail: [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: SIP/zvonobot-00000012: Not in SFU mode [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining softmix technology [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Recorder/ARI-00000003;2: [Aug 18 10:33:50] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50397': 0x7f0c8402b380 created [Aug 18 10:33:50] DEBUG[13222] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:50] DEBUG[13222] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Recorder/ARI-00000003;2: [Aug 18 10:33:50] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50397' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:50] DEBUG[13222] bridge_softmix.c: Recorder/ARI-00000003;2: Not in SFU mode [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology start [Aug 18 10:33:50] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology destructor [Aug 18 10:33:50] VERBOSE[13221] dial.c: UnicastRTP/127.0.0.1:50397-0x7f0c28011240 answered [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:50] DEBUG[13223] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: starting mixing thread [Aug 18 10:33:50] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:33:50] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP ooh, format changed from none to alaw [Aug 18 10:33:50] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP starting transmission [Aug 18 10:33:50] DEBUG[13217] res_stasis_playback.c: 1629282830.63: Sending play(sound:silence/2) command [Aug 18 10:33:50] VERBOSE[13221] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50397-0x7f0c28011240 [Aug 18 10:33:50] DEBUG[13217] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:50] DEBUG[13217] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13224] channel.c: Channel Announcer/ARI-00000004;1 setting write format path: gsm -> slin [Aug 18 10:33:50] DEBUG[13224] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:50] VERBOSE[13208] res_rtp_asterisk.c: 0x7f0c100223b0 -- Strict RTP switching to RTP target address 178.62.121.41:10788 as source [Aug 18 10:33:50] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:33:50] DEBUG[13221] stasis/app.c: Channel 'robot_212983' is 2 interested in calls_0 [Aug 18 10:33:50] VERBOSE[13224] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:50] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:50] DEBUG[13225] http.c: HTTP opening session. Top level [Aug 18 10:33:50] DEBUG[13225] http.c: HTTP Request URI is /ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee/addChannel?channel=1629282830.62%2Crobot_212983 [Aug 18 10:33:50] DEBUG[13225] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:50] DEBUG[13225] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:50] DEBUG[13225] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee/addChannel] with handler [ari] len 3 [Aug 18 10:33:50] DEBUG[13225] http.c: Match made with [ari] [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Finding handler for bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee/addChannel [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Finding handler for bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Finding handler for 3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:50] DEBUG[13225] res_ari.c: No explicit handler found for 3f704757-87e2-45e5-8aa9-92ed6ea9feee. Using wildcard bridgeId. [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Finding handler for addChannel [Aug 18 10:33:50] DEBUG[13225] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:50] DEBUG[13225] stasis/control.c: 1629282830.62: Sending channel add_to_bridge command [Aug 18 10:33:50] DEBUG[13214] bridge_roles.c: Roles did not exist on channel Snoop/212983-00000001 [Aug 18 10:33:50] DEBUG[13214] stasis/control.c: 1629282830.62: Adding to bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:33:50] DEBUG[13214] stasis/app.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' is 1 interested in calls_0 [Aug 18 10:33:50] DEBUG[13226] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c2000f4c0(Snoop/212983-00000001) is joining [Aug 18 10:33:50] DEBUG[13226] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: pushing 0x7f0c2000f4c0(Snoop/212983-00000001) [Aug 18 10:33:50] VERBOSE[13226] bridge_channel.c: Channel Snoop/212983-00000001 joined 'simple_bridge' stasis-bridge <3f704757-87e2-45e5-8aa9-92ed6ea9feee> [Aug 18 10:33:50] DEBUG[13226] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' can not use native RTP bridge as two channels are required [Aug 18 10:33:50] DEBUG[13226] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13226] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13226] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13226] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13226] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee is already using the new technology. [Aug 18 10:33:50] DEBUG[13226] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c2000f4c0(Snoop/212983-00000001) is joining simple_bridge technology [Aug 18 10:33:50] DEBUG[13225] stasis/control.c: robot_212983: Sending channel add_to_bridge command [Aug 18 10:33:50] DEBUG[13214] stasis/app.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' is 2 interested in calls_0 [Aug 18 10:33:50] DEBUG[13221] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50397-0x7f0c28011240 [Aug 18 10:33:50] DEBUG[13221] stasis/control.c: robot_212983: Adding to bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:33:50] DEBUG[13221] stasis/app.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' is 3 interested in calls_0 [Aug 18 10:33:50] DEBUG[13227] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) is joining [Aug 18 10:33:50] DEBUG[13227] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: pushing 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) [Aug 18 10:33:50] VERBOSE[13227] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50397-0x7f0c28011240 joined 'simple_bridge' stasis-bridge <3f704757-87e2-45e5-8aa9-92ed6ea9feee> [Aug 18 10:33:50] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50397-0x7f0c28011240 - start 1629282830.505806 answer 1629282830.509264 end 1629282830.723776 dur 0.217 bill 0.214 dispo ANSWERED [Aug 18 10:33:50] DEBUG[13227] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee'. Checking compatability for channels 'Snoop/212983-00000001' and 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240' [Aug 18 10:33:50] DEBUG[13227] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' can not use native RTP bridge as could not get details [Aug 18 10:33:50] DEBUG[13227] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:50] DEBUG[13227] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13227] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:50] DEBUG[13227] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:50] DEBUG[13227] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee is already using the new technology. [Aug 18 10:33:50] DEBUG[13227] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) is joining simple_bridge technology [Aug 18 10:33:50] DEBUG[13227] channel.c: Channel UnicastRTP/127.0.0.1:50397-0x7f0c28011240 setting read format path: slin16 -> slin16 [Aug 18 10:33:50] DEBUG[13227] channel.c: Channel Snoop/212983-00000001 setting write format path: slin16 -> slin [Aug 18 10:33:50] DEBUG[13227] channel.c: Channel Snoop/212983-00000001 setting read format path: slin -> slin16 [Aug 18 10:33:50] DEBUG[13227] channel.c: Channel UnicastRTP/127.0.0.1:50397-0x7f0c28011240 setting write format path: slin16 -> slin16 [Aug 18 10:33:50] DEBUG[13221] stasis/app.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' is 4 interested in calls_0 [Aug 18 10:33:50] DEBUG[13225] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:50] DEBUG[13225] http.c: HTTP closing session. Top level [Aug 18 10:33:50] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP ooh, format changed from none to slin16 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41960c20;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 657733984 657733984 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18326 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41960c20;received=159.65.48.104 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3725d74e [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 657733984 657733984 IN IP4 178.62.121.41 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18326 RTP/AVP 0 8 101 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as3725d74e [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Stopping retransmission on '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Got SDP version 657733984 and unique parts [root 657733984 IN IP4 178.62.121.41] [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 657733984 657733984 IN IP4 178.62.121.41... OK. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:51] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:51] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:51] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:51] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) ICE set role failed; no ice instance [Aug 18 10:33:51] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) RTCP setting address on RTP instance [Aug 18 10:33:51] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c3401d7e0 -- Strict RTP learning after remote address set to: 178.62.121.41:18326 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:18326 [Aug 18 10:33:51] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0036cc8) from 0x7f0c147e2330 to 0x7f0c3401c268 [Aug 18 10:33:51] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00696e8) from 0x7f0c147e2330 to 0x7f0c3401c268 [Aug 18 10:33:51] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0076bb8) from 0x7f0c147e2330 to 0x7f0c3401c268 [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) RTCP ignoring duplicate property [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:51] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000016 setting read format path: alaw -> alaw [Aug 18 10:33:51] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000016 setting write format path: alaw -> alaw [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS - ast_rtp_activate rtp=0x7f0c3401d7e0 - setup and perform DTLS' [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401d7e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401d7e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:51] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:51] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:51] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Strict routing enforced for session 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:51] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:51] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65e81ae1 Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Contact: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Session timer started: 14 - 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 1768000ms [Aug 18 10:33:51] VERBOSE[12991] dial.c: SIP/zvonobot-00000016 answered [Aug 18 10:33:51] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:51] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:51] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:51] VERBOSE[12991] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000016 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:51] DEBUG[12991] stasis/app.c: Channel '212986' is 2 interested in calls_0 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:51] DEBUG[13228] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13228] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:51] DEBUG[13228] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13228] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13228] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13228] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13228] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13228] stasis.c: Creating topic. name: bridge:90245cdf-0ee9-4414-b99e-c22349f119a2, detail: [Aug 18 10:33:51] DEBUG[13228] stasis.c: Topic 'bridge:90245cdf-0ee9-4414-b99e-c22349f119a2': 0x7f0c7c0068a0 created [Aug 18 10:33:51] DEBUG[13228] stasis.c: Creating topic. name: cache:77/bridge:90245cdf-0ee9-4414-b99e-c22349f119a2, detail: [Aug 18 10:33:51] DEBUG[13228] stasis.c: Topic 'cache:77/bridge:90245cdf-0ee9-4414-b99e-c22349f119a2': 0x7f0c7c017490 created [Aug 18 10:33:51] DEBUG[13228] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:51] DEBUG[13228] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13228] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13228] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:51] DEBUG[13228] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13228] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology constructor [Aug 18 10:33:51] DEBUG[13228] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology start [Aug 18 10:33:51] DEBUG[13228] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:51] DEBUG[13228] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13229] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13229] http.c: HTTP Request URI is /ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/addChannel?channel=212986 [Aug 18 10:33:51] DEBUG[13229] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13229] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13229] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/addChannel] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13229] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Finding handler for bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/addChannel [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Finding handler for 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13229] res_ari.c: No explicit handler found for 90245cdf-0ee9-4414-b99e-c22349f119a2. Using wildcard bridgeId. [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Finding handler for addChannel [Aug 18 10:33:51] DEBUG[13229] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:51] DEBUG[13229] stasis/control.c: 212986: Sending channel add_to_bridge command [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517;received=159.65.48.104 From: ;tag=as6af53e10 To: ;tag=as316e0345 Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517;received=159.65.48.104 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6af53e10 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as316e0345 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 (Checking To) --From tag as6af53e10 --To-tag as316e0345 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Stopping retransmission on '72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:51] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:51] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117072@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10f58517 Max-Forwards: 70 From: ;tag=as6af53e10 To: ;tag=as316e0345 Contact: Call-ID: 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:51] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:51] VERBOSE[12879] dial.c: SIP/zvonobot-00000003 is busy [Aug 18 10:33:51] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000003 - start 1629282822.111840 answer 0.000000 end 1629282831.124712 dur 9.012 bill 1629282831.124 dispo BUSY [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:51] DEBUG[12879] channel.c: Channel 0x7f0c1c016880 'SIP/zvonobot-00000003' hanging up. Refs: 2 [Aug 18 10:33:51] DEBUG[12879] chan_sip.c: Hangup call SIP/zvonobot-00000003, SIP callid 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:51] DEBUG[12879] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:51] DEBUG[12879] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:51] DEBUG[12879] channel.c: Channel 0x7f0c1c016880 'SIP/zvonobot-00000003' destroying [Aug 18 10:33:51] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282831.66, detail: [Aug 18 10:33:51] DEBUG[20545] stasis.c: Topic 'channel:1629282831.66': 0x7f0c3004b1a0 created [Aug 18 10:33:51] DEBUG[20545] stasis.c: Creating topic. name: cache:78/channel:1629282831.66, detail: [Aug 18 10:33:51] DEBUG[20545] stasis.c: Topic 'cache:78/channel:1629282831.66': 0x7f0c300809f0 created [Aug 18 10:33:51] DEBUG[20545] stasis.c: Destroying topic. name: cache:78/channel:1629282831.66, detail: [Aug 18 10:33:51] DEBUG[20545] stasis.c: Topic 'cache:78/channel:1629282831.66': 0x7f0c300809f0 destroyed [Aug 18 10:33:51] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282831.66, detail: [Aug 18 10:33:51] DEBUG[20545] stasis.c: Topic 'channel:1629282831.66': 0x7f0c3004b1a0 destroyed [Aug 18 10:33:51] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000003', '', 'AppDial2', '(Outgoing Line)', 9, 0, 'BUSY', 3, '', '212968', '')] [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:33:51] DEBUG[20620] stasis/app.c: channel '212968': is 0 interested in calls_0 [Aug 18 10:33:51] DEBUG[20620] stasis/app.c: channel '212968' unsubscribed from calls_0 [Aug 18 10:33:51] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:51] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:51] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:51] DEBUG[20620] stasis.c: Destroying topic. name: cache:10/channel:212968, detail: [Aug 18 10:33:51] DEBUG[20620] stasis.c: Topic 'cache:10/channel:212968': 0x7f0c1c07cfc0 destroyed [Aug 18 10:33:51] DEBUG[20620] stasis.c: Destroying topic. name: channel:212968, detail: [Aug 18 10:33:51] DEBUG[20620] stasis.c: Topic 'channel:212968': 0x7f0c1c018590 destroyed [Aug 18 10:33:51] DEBUG[13230] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13230] http.c: HTTP Request URI is /ari/channels/212968 [Aug 18 10:33:51] DEBUG[13230] http.c: match request [ari/channels/212968] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13230] http.c: match request [ari/channels/212968] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13230] http.c: match request [ari/channels/212968] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13230] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Finding handler for channels/212968 [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Finding handler for channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Finding handler for 212968 [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking channels create: Didn't match 212968 [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13230] res_ari.c: Checking channels externalMedia: Didn't match 212968 [Aug 18 10:33:51] DEBUG[13230] res_ari.c: No explicit handler found for 212968. Using wildcard channelId. [Aug 18 10:33:51] DEBUG[13230] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:51] DEBUG[13230] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[12991] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000016 [Aug 18 10:33:51] DEBUG[12991] stasis/control.c: 212986: Adding to bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:33:51] DEBUG[12991] stasis/app.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' is 1 interested in calls_0 [Aug 18 10:33:51] DEBUG[13232] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is joining [Aug 18 10:33:51] DEBUG[13232] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pushing 0x7f0c30064e70(SIP/zvonobot-00000016) [Aug 18 10:33:51] VERBOSE[13232] bridge_channel.c: Channel SIP/zvonobot-00000016 joined 'simple_bridge' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:51] DEBUG[13232] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 is already using the new technology. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is joining simple_bridge technology [Aug 18 10:33:51] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTP changing ssrc from 84433688 to 1119016996 due to a source change [Aug 18 10:33:51] DEBUG[12991] stasis/app.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' is 2 interested in calls_0 [Aug 18 10:33:51] DEBUG[13229] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:51] DEBUG[13229] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13233] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13233] http.c: HTTP Request URI is /ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/record?name=212986_yjBHvJSYCdJpnHGsVZbGqJlgkPNdXnIn&format=wav [Aug 18 10:33:51] DEBUG[13233] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/record] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13233] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/record] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13233] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/record] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13233] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Finding handler for bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Finding handler for 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13233] res_ari.c: No explicit handler found for 90245cdf-0ee9-4414-b99e-c22349f119a2. Using wildcard bridgeId. [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Finding handler for record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:51] DEBUG[13233] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:51] DEBUG[13233] stasis.c: Creating topic. name: channel:1629282831.67, detail: [Aug 18 10:33:51] DEBUG[13233] stasis.c: Topic 'channel:1629282831.67': 0x7f0c8c018b60 created [Aug 18 10:33:51] DEBUG[13233] stasis.c: Creating topic. name: cache:79/channel:1629282831.67, detail: [Aug 18 10:33:51] DEBUG[13233] stasis.c: Topic 'cache:79/channel:1629282831.67': 0x7f0c8c030cf0 created [Aug 18 10:33:51] DEBUG[13233] channel.c: Channel 0x7f0c8c02ea90 'Recorder/ARI-00000005;1' allocated [Aug 18 10:33:51] DEBUG[13233] stasis.c: Creating topic. name: channel:1629282831.68, detail: [Aug 18 10:33:51] DEBUG[13233] stasis.c: Topic 'channel:1629282831.68': 0x7f0c8c038f60 created [Aug 18 10:33:51] DEBUG[13233] stasis.c: Creating topic. name: cache:80/channel:1629282831.68, detail: [Aug 18 10:33:51] DEBUG[13233] stasis.c: Topic 'cache:80/channel:1629282831.68': 0x7f0c8c01ff90 created [Aug 18 10:33:51] DEBUG[13233] channel.c: Channel 0x7f0c8c037210 'Recorder/ARI-00000005;2' allocated [Aug 18 10:33:51] DEBUG[13233] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:51] DEBUG[13234] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is joining [Aug 18 10:33:51] DEBUG[13234] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pushing 0x7f0c8c037dd0(Recorder/ARI-00000005;2) [Aug 18 10:33:51] DEBUG[13234] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:51] VERBOSE[13234] bridge_channel.c: Channel Recorder/ARI-00000005;2 joined 'simple_bridge' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:51] DEBUG[13234] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2'. Checking compatability for channels 'SIP/zvonobot-00000016' and 'Recorder/ARI-00000005;2' [Aug 18 10:33:51] DEBUG[13234] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as could not get details [Aug 18 10:33:51] DEBUG[13234] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13234] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13234] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13234] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13234] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 is already using the new technology. [Aug 18 10:33:51] DEBUG[13234] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is joining simple_bridge technology [Aug 18 10:33:51] DEBUG[13234] channel.c: Channel Recorder/ARI-00000005;2 setting read format path: slin -> slin [Aug 18 10:33:51] DEBUG[13234] channel.c: Channel SIP/zvonobot-00000016 setting write format path: slin -> alaw [Aug 18 10:33:51] DEBUG[13234] channel.c: Channel SIP/zvonobot-00000016 setting read format path: alaw -> slin [Aug 18 10:33:51] DEBUG[13234] channel.c: Channel Recorder/ARI-00000005;2 setting write format path: slin -> slin [Aug 18 10:33:51] DEBUG[13233] res_stasis_recording.c: 1629282831.67: Sending record(212986_yjBHvJSYCdJpnHGsVZbGqJlgkPNdXnIn.wav) command [Aug 18 10:33:51] DEBUG[13235] app.c: play_and_record: , /var/spool/asterisk/recording/212986_yjBHvJSYCdJpnHGsVZbGqJlgkPNdXnIn, 'wav' [Aug 18 10:33:51] DEBUG[13235] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:51] DEBUG[13233] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:51] VERBOSE[13235] app.c: x=0, open writing: /var/spool/asterisk/recording/212986_yjBHvJSYCdJpnHGsVZbGqJlgkPNdXnIn format: wav, 0x7f0c94029a60 [Aug 18 10:33:51] DEBUG[13233] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13236] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13236] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:51] DEBUG[13236] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13236] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13236] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13236] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13236] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13236] stasis.c: Creating topic. name: bridge:8b092052-108a-4921-8aad-1aecb4e2c824, detail: [Aug 18 10:33:51] DEBUG[13236] stasis.c: Topic 'bridge:8b092052-108a-4921-8aad-1aecb4e2c824': 0x7f0c90021250 created [Aug 18 10:33:51] DEBUG[13236] stasis.c: Creating topic. name: cache:81/bridge:8b092052-108a-4921-8aad-1aecb4e2c824, detail: [Aug 18 10:33:51] DEBUG[13236] stasis.c: Topic 'cache:81/bridge:8b092052-108a-4921-8aad-1aecb4e2c824': 0x7f0c900160a0 created [Aug 18 10:33:51] DEBUG[13236] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' can not use native RTP bridge as two channels are required [Aug 18 10:33:51] DEBUG[13236] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13236] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13236] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:51] DEBUG[13236] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13236] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: calling simple_bridge technology constructor [Aug 18 10:33:51] DEBUG[13236] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: calling simple_bridge technology start [Aug 18 10:33:51] DEBUG[13236] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:51] DEBUG[13236] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13237] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13237] http.c: HTTP Request URI is /ari/channels/212986/snoop?app=calls_0&spy=in [Aug 18 10:33:51] DEBUG[13237] http.c: match request [ari/channels/212986/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13237] http.c: match request [ari/channels/212986/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13237] http.c: match request [ari/channels/212986/snoop] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13237] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Finding handler for channels/212986/snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Finding handler for channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Finding handler for 212986 [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channels create: Didn't match 212986 [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channels externalMedia: Didn't match 212986 [Aug 18 10:33:51] DEBUG[13237] res_ari.c: No explicit handler found for 212986. Using wildcard channelId. [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Finding handler for snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:51] DEBUG[13237] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:51] DEBUG[13237] stasis.c: Creating topic. name: channel:1629282831.69, detail: [Aug 18 10:33:51] DEBUG[13237] stasis.c: Topic 'channel:1629282831.69': 0x7f0ca802e8b0 created [Aug 18 10:33:51] DEBUG[13237] stasis.c: Creating topic. name: cache:82/channel:1629282831.69, detail: [Aug 18 10:33:51] DEBUG[13237] stasis.c: Topic 'cache:82/channel:1629282831.69': 0x7f0ca800ea90 created [Aug 18 10:33:51] DEBUG[13237] channel.c: Channel 0x7f0ca8007d50 'Snoop/212986-00000002' allocated [Aug 18 10:33:51] DEBUG[13232] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2'. Checking compatability for channels 'SIP/zvonobot-00000016' and 'Recorder/ARI-00000005;2' [Aug 18 10:33:51] DEBUG[13232] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as channel 'SIP/zvonobot-00000016' has features which prevent it [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13232] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13232] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 is already using the new technology. [Aug 18 10:33:51] DEBUG[13237] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:51] DEBUG[13237] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13241] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13241] http.c: HTTP Request URI is /ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play?media=sound%3Asilence%2F2 [Aug 18 10:33:51] DEBUG[13241] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13241] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13241] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13241] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Finding handler for bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Finding handler for 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13238] stasis/app.c: Channel '1629282831.69' is 1 interested in calls_0 [Aug 18 10:33:51] DEBUG[13241] res_ari.c: No explicit handler found for 90245cdf-0ee9-4414-b99e-c22349f119a2. Using wildcard bridgeId. [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Finding handler for play [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 457 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 457 [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:51] DEBUG[13241] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:51] DEBUG[13241] stasis.c: Creating topic. name: channel:1629282831.70, detail: [Aug 18 10:33:51] DEBUG[13241] stasis.c: Topic 'channel:1629282831.70': 0x7f0c9802a530 created [Aug 18 10:33:51] DEBUG[13241] stasis.c: Creating topic. name: cache:83/channel:1629282831.70, detail: [Aug 18 10:33:51] DEBUG[13241] stasis.c: Topic 'cache:83/channel:1629282831.70': 0x7f0c98023d90 created [Aug 18 10:33:51] DEBUG[13241] channel.c: Channel 0x7f0c9802bfd0 'Announcer/ARI-00000006;1' allocated [Aug 18 10:33:51] DEBUG[13241] stasis.c: Creating topic. name: channel:1629282831.71, detail: [Aug 18 10:33:51] DEBUG[13241] stasis.c: Topic 'channel:1629282831.71': 0x7f0c98036be0 created [Aug 18 10:33:51] DEBUG[13241] stasis.c: Creating topic. name: cache:84/channel:1629282831.71, detail: [Aug 18 10:33:51] DEBUG[13241] stasis.c: Topic 'cache:84/channel:1629282831.71': 0x7f0c98037660 created [Aug 18 10:33:51] DEBUG[13241] channel.c: Channel 0x7f0c98034ab0 'Announcer/ARI-00000006;2' allocated [Aug 18 10:33:51] DEBUG[13241] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:51] DEBUG[13241] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000006;1' [Aug 18 10:33:51] DEBUG[13243] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13245] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c9802b450(Announcer/ARI-00000006;2) is joining [Aug 18 10:33:51] DEBUG[13243] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212986&app=calls_0&format=slin16&external_host=127.0.0.1%3A50291 [Aug 18 10:33:51] DEBUG[13243] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13243] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13243] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13243] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13245] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pushing 0x7f0c9802b450(Announcer/ARI-00000006;2) [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Finding handler for channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:51] DEBUG[13245] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:51] VERBOSE[13245] bridge_channel.c: Channel Announcer/ARI-00000006;2 joined 'simple_bridge' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13243] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:51] DEBUG[13243] netsock2.c: Splitting '127.0.0.1:50291' into... [Aug 18 10:33:51] DEBUG[13243] netsock2.c: ...host '127.0.0.1' and port '50291'. [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13243] netsock2.c: Splitting '127.0.0.1:50291' into... [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13243] netsock2.c: ...host '127.0.0.1' and port '50291'. [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13245] bridge.c: Chose bridge technology softmix [Aug 18 10:33:51] VERBOSE[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: switching from simple_bridge technology to softmix [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling softmix technology constructor [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: moving 0x7f0c30064e70(SIP/zvonobot-00000016) to dummy bridge temporarily [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: moving 0x7f0c8c037dd0(Recorder/ARI-00000005;2) to dummy bridge temporarily [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is leaving simple_bridge technology (dummy) [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:51] DEBUG[13243] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology stop [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c9802b450(Announcer/ARI-00000006;2) is joining softmix technology [Aug 18 10:33:51] DEBUG[13243] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca0023720' [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Announcer/ARI-00000006;2: [Aug 18 10:33:51] DEBUG[13245] channel.c: Channel Announcer/ARI-00000006;2 setting write format path: slin -> slin [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Announcer/ARI-00000006;2: [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) RTP allocated port 18614 [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Announcer/ARI-00000006;2: Not in SFU mode [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) ICE creating session 127.0.0.1:18614 (18614) [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is joining softmix technology [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: SIP/zvonobot-00000016: [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) ICE create [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: SIP/zvonobot-00000016: [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: SIP/zvonobot-00000016: Not in SFU mode [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is joining softmix technology [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Recorder/ARI-00000005;2: [Aug 18 10:33:51] DEBUG[13245] channel.c: Channel Recorder/ARI-00000005;2 setting write format path: slin -> slin [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) ICE add system candidates [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:51] DEBUG[13245] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:51] DEBUG[13243] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Recorder/ARI-00000005;2: [Aug 18 10:33:51] DEBUG[13243] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) ICE add candidate: 159.65.48.104:18614, 2130706431 [Aug 18 10:33:51] DEBUG[13245] bridge_softmix.c: Recorder/ARI-00000005;2: Not in SFU mode [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling softmix technology start [Aug 18 10:33:51] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology destructor [Aug 18 10:33:51] DEBUG[13243] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:51] DEBUG[13243] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:51] DEBUG[13243] res_rtp_asterisk.c: (0x7f0ca0023720) ICE add candidate: 10.131.0.10:18614, 2130706431 [Aug 18 10:33:51] DEBUG[13243] rtp_engine.c: RTP instance '0x7f0ca0023720' is setup and ready to go [Aug 18 10:33:51] DEBUG[13243] stasis.c: Creating topic. name: channel:robot_212986, detail: [Aug 18 10:33:51] DEBUG[13243] stasis.c: Topic 'channel:robot_212986': 0x7f0ca002f0e0 created [Aug 18 10:33:51] DEBUG[13243] stasis.c: Creating topic. name: cache:85/channel:robot_212986, detail: [Aug 18 10:33:51] DEBUG[13243] stasis.c: Topic 'cache:85/channel:robot_212986': 0x7f0ca002e2f0 created [Aug 18 10:33:51] DEBUG[13246] bridge_softmix.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: starting mixing thread [Aug 18 10:33:51] DEBUG[13241] res_stasis_playback.c: 1629282831.70: Sending play(sound:silence/2) command [Aug 18 10:33:51] DEBUG[13232] audiohook.c: Audiohook 0x7f0ca802f950 has stale audio in its factories. Flushing them both [Aug 18 10:33:51] DEBUG[13241] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:51] DEBUG[13241] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTP ooh, format changed from none to alaw [Aug 18 10:33:51] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTCP starting transmission [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:51] VERBOSE[13232] res_rtp_asterisk.c: 0x7f0c3401d7e0 -- Strict RTP switching to RTP target address 178.62.121.41:18326 as source [Aug 18 10:33:51] DEBUG[13232] audiohook.c: Audiohook 0x7f0ca802f950 has stale audio in its factories. Flushing them both [Aug 18 10:33:51] DEBUG[13247] channel.c: Channel Announcer/ARI-00000006;1 setting write format path: gsm -> slin [Aug 18 10:33:51] DEBUG[13243] channel.c: Channel 0x7f0ca002fd30 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720' allocated [Aug 18 10:33:51] DEBUG[13243] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:51] DEBUG[13247] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:51] VERBOSE[13243] res_rtp_asterisk.c: 0x7f0ca0028fe0 -- Strict RTP learning after remote address set to: 127.0.0.1:50291 [Aug 18 10:33:51] VERBOSE[13247] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:51] DEBUG[13243] res_stasis.c: calls_0: Subscribing to robot_212986 [Aug 18 10:33:51] DEBUG[13243] stasis/app.c: Channel 'robot_212986' is 1 interested in calls_0 [Aug 18 10:33:51] VERBOSE[13248] dial.c: Called 127.0.0.1:50291 [Aug 18 10:33:51] VERBOSE[13248] dial.c: UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 answered [Aug 18 10:33:51] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50291 [Aug 18 10:33:51] VERBOSE[13248] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:51] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50291 - state 2 (In use) [Aug 18 10:33:51] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50291, detail: [Aug 18 10:33:51] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50291': 0x7f0c84040890 created [Aug 18 10:33:51] DEBUG[13248] stasis/app.c: Channel 'robot_212986' is 2 interested in calls_0 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:51] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:51] DEBUG[13243] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:51] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50291' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:51] DEBUG[13243] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13249] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13249] http.c: HTTP Request URI is /ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824/addChannel?channel=1629282831.69%2Crobot_212986 [Aug 18 10:33:51] DEBUG[13249] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13249] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13249] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824/addChannel] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13249] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Finding handler for bridges/8b092052-108a-4921-8aad-1aecb4e2c824/addChannel [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Finding handler for bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Finding handler for 8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13249] res_ari.c: No explicit handler found for 8b092052-108a-4921-8aad-1aecb4e2c824. Using wildcard bridgeId. [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Finding handler for addChannel [Aug 18 10:33:51] DEBUG[13249] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:51] DEBUG[13249] stasis/control.c: 1629282831.69: Sending channel add_to_bridge command [Aug 18 10:33:51] DEBUG[13238] bridge_roles.c: Roles did not exist on channel Snoop/212986-00000002 [Aug 18 10:33:51] DEBUG[13238] stasis/control.c: 1629282831.69: Adding to bridge 8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:33:51] DEBUG[13238] stasis/app.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' is 1 interested in calls_0 [Aug 18 10:33:51] DEBUG[13250] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0c9c021fe0(Snoop/212986-00000002) is joining [Aug 18 10:33:51] DEBUG[13250] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: pushing 0x7f0c9c021fe0(Snoop/212986-00000002) [Aug 18 10:33:51] VERBOSE[13250] bridge_channel.c: Channel Snoop/212986-00000002 joined 'simple_bridge' stasis-bridge <8b092052-108a-4921-8aad-1aecb4e2c824> [Aug 18 10:33:51] DEBUG[13250] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' can not use native RTP bridge as two channels are required [Aug 18 10:33:51] DEBUG[13250] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13250] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13250] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13250] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13250] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824 is already using the new technology. [Aug 18 10:33:51] DEBUG[13250] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0c9c021fe0(Snoop/212986-00000002) is joining simple_bridge technology [Aug 18 10:33:51] DEBUG[13249] stasis/control.c: robot_212986: Sending channel add_to_bridge command [Aug 18 10:33:51] DEBUG[13238] stasis/app.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' is 2 interested in calls_0 [Aug 18 10:33:51] DEBUG[13248] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 [Aug 18 10:33:51] DEBUG[13248] stasis/control.c: robot_212986: Adding to bridge 8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:33:51] DEBUG[13248] stasis/app.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' is 3 interested in calls_0 [Aug 18 10:33:51] DEBUG[13251] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) is joining [Aug 18 10:33:51] DEBUG[13251] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: pushing 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) [Aug 18 10:33:51] VERBOSE[13251] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 joined 'simple_bridge' stasis-bridge <8b092052-108a-4921-8aad-1aecb4e2c824> [Aug 18 10:33:51] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 - start 1629282831.325580 answer 1629282831.330299 end 1629282831.533499 dur 0.207 bill 0.203 dispo ANSWERED [Aug 18 10:33:51] DEBUG[13251] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824'. Checking compatability for channels 'Snoop/212986-00000002' and 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720' [Aug 18 10:33:51] DEBUG[13251] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' can not use native RTP bridge as could not get details [Aug 18 10:33:51] DEBUG[13251] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:51] DEBUG[13251] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13251] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:51] DEBUG[13251] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:51] DEBUG[13251] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824 is already using the new technology. [Aug 18 10:33:51] DEBUG[13251] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) is joining simple_bridge technology [Aug 18 10:33:51] DEBUG[13251] channel.c: Channel UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 setting read format path: slin16 -> slin16 [Aug 18 10:33:51] DEBUG[13251] channel.c: Channel Snoop/212986-00000002 setting write format path: slin16 -> slin [Aug 18 10:33:51] DEBUG[13251] channel.c: Channel Snoop/212986-00000002 setting read format path: slin -> slin16 [Aug 18 10:33:51] DEBUG[13251] channel.c: Channel UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 setting write format path: slin16 -> slin16 [Aug 18 10:33:51] DEBUG[13248] stasis/app.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' is 4 interested in calls_0 [Aug 18 10:33:51] DEBUG[13249] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:51] DEBUG[13249] http.c: HTTP closing session. Top level [Aug 18 10:33:51] DEBUG[13251] res_rtp_asterisk.c: (0x7f0ca0023720) RTP ooh, format changed from none to slin16 [Aug 18 10:33:51] DEBUG[13254] http.c: HTTP opening session. Top level [Aug 18 10:33:51] DEBUG[13254] http.c: HTTP Request URI is /ari/channels/213014?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117026&callerId=74950493843 [Aug 18 10:33:51] DEBUG[13254] http.c: match request [ari/channels/213014] with handler [httpstatus] len 10 [Aug 18 10:33:51] DEBUG[13254] http.c: match request [ari/channels/213014] with handler [phoneprov] len 9 [Aug 18 10:33:51] DEBUG[13254] http.c: match request [ari/channels/213014] with handler [ari] len 3 [Aug 18 10:33:51] DEBUG[13254] http.c: Match made with [ari] [Aug 18 10:33:51] DEBUG[13254] http.c: HTTP consuming request body [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Finding handler for channels/213014 [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Finding handler for channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Finding handler for 213014 [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking channels create: Didn't match 213014 [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:51] DEBUG[13254] res_ari.c: Checking channels externalMedia: Didn't match 213014 [Aug 18 10:33:51] DEBUG[13254] res_ari.c: No explicit handler found for 213014. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: Allocating new SIP dialog for 548d5f91381a62e82e842c9d291a2223@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13254] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c00b650' [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) RTP allocated port 13724 [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE creating session 0.0.0.0:13724 (13724) [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE create [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE add system candidates [Aug 18 10:33:52] DEBUG[13254] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13254] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE add candidate: 159.65.48.104:13724, 2130706431 [Aug 18 10:33:52] DEBUG[13254] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13254] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE add candidate: 10.131.0.10:13724, 2130706431 [Aug 18 10:33:52] DEBUG[13254] rtp_engine.c: RTP instance '0x7f0c1c00b650' is setup and ready to go [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE stopped [Aug 18 10:33:52] DEBUG[13254] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13254] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13254] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13254] res_rtp_asterisk.c: (0x7f0c1c00b650) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13254] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13254] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: SIP call-id changed from '548d5f91381a62e82e842c9d291a2223@127.0.1.1:5060' to '321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13254] stasis.c: Creating topic. name: channel:213014, detail: [Aug 18 10:33:52] DEBUG[13254] stasis.c: Topic 'channel:213014': 0x7f0c1c01a790 created [Aug 18 10:33:52] DEBUG[13254] stasis.c: Creating topic. name: cache:86/channel:213014, detail: [Aug 18 10:33:52] DEBUG[13254] stasis.c: Topic 'cache:86/channel:213014': 0x7f0c1c043090 created [Aug 18 10:33:52] DEBUG[13254] channel.c: Channel 0x7f0c1c041e00 'SIP/zvonobot-00000032' allocated [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13254] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13254] res_stasis.c: calls_0: Subscribing to 213014 [Aug 18 10:33:52] DEBUG[13254] stasis/app.c: Channel '213014' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13254] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Destroying SIP dialog 72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '72accbaf2ae6a18b60808a0d79c893ff@159.65.48.104:5060' Method: INVITE [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS stop [Aug 18 10:33:52] DEBUG[13254] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00f0c0) ICE RTP transport deallocating [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c1c00f0c0' [Aug 18 10:33:52] DEBUG[13256] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Outgoing Call for 79821117026 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13256] http.c: HTTP Request URI is /ari/channels/213015?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117025&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13256] http.c: match request [ari/channels/213015] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13256] http.c: match request [ari/channels/213015] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13256] http.c: match request [ari/channels/213015] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13256] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13256] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Finding handler for channels/213015 [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Finding handler for 213015 [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking channels create: Didn't match 213015 [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13256] res_ari.c: Checking channels externalMedia: Didn't match 213015 [Aug 18 10:33:52] DEBUG[13256] res_ari.c: No explicit handler found for 213015. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[13255] chan_sip.c: Audio is at 13724 [Aug 18 10:33:52] VERBOSE[13255] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13255] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13255] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Initializing initreq for method INVITE - callid 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117026@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5443b9 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 3 [ 52]: From: ;tag=as57703f31 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 6 [ 60]: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13255] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117026@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5443b9 Max-Forwards: 70 From: ;tag=as57703f31 To: Contact: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 665003889 665003889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13724 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:52] DEBUG[13255] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[13255] dial.c: Called zvonobot/79821117026 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5443b9;received=159.65.48.104 From: ;tag=as57703f31 To: ;tag=as1be323d9 Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="175aa82f" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5443b9;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57703f31 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1be323d9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="175aa82f" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 (Checking To) --From tag as57703f31 --To-tag as1be323d9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117026@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5443b9 Max-Forwards: 70 From: ;tag=as57703f31 To: ;tag=as1be323d9 Contact: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 13724 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117026@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b9cf791 Max-Forwards: 70 From: ;tag=as57703f31 To: Contact: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117026@178.62.121.41", nonce="175aa82f", response="10b8c439fe10c34589cdf4b8052ba852" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 665003889 665003890 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13724 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b9cf791;received=159.65.48.104 From: ;tag=as57703f31 To: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b9cf791;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57703f31 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 (Checking To) --From tag as57703f31 --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #13 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: Allocating new SIP dialog for 075ed4fe213158280b2c9fbb7b8ceea8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13256] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c24032c40' [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) RTP allocated port 16466 [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE creating session 0.0.0.0:16466 (16466) [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE create [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE add system candidates [Aug 18 10:33:52] DEBUG[13256] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13256] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13260] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13260] http.c: HTTP Request URI is /ari/channels/213016?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117024&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13260] http.c: match request [ari/channels/213016] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE add candidate: 159.65.48.104:16466, 2130706431 [Aug 18 10:33:52] DEBUG[13256] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13256] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13260] http.c: match request [ari/channels/213016] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13260] http.c: match request [ari/channels/213016] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13260] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13260] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Finding handler for channels/213016 [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Finding handler for 213016 [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking channels create: Didn't match 213016 [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13260] res_ari.c: Checking channels externalMedia: Didn't match 213016 [Aug 18 10:33:52] DEBUG[13260] res_ari.c: No explicit handler found for 213016. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE add candidate: 10.131.0.10:16466, 2130706431 [Aug 18 10:33:52] DEBUG[13256] rtp_engine.c: RTP instance '0x7f0c24032c40' is setup and ready to go [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) ICE stopped [Aug 18 10:33:52] DEBUG[13256] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13256] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13256] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13256] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13256] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13256] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: SIP call-id changed from '075ed4fe213158280b2c9fbb7b8ceea8@127.0.1.1:5060' to '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13256] stasis.c: Creating topic. name: channel:213015, detail: [Aug 18 10:33:52] DEBUG[13256] stasis.c: Topic 'channel:213015': 0x7f0c2403f710 created [Aug 18 10:33:52] DEBUG[13256] stasis.c: Creating topic. name: cache:87/channel:213015, detail: [Aug 18 10:33:52] DEBUG[13256] stasis.c: Topic 'cache:87/channel:213015': 0x7f0c2403f310 created [Aug 18 10:33:52] DEBUG[13256] channel.c: Channel 0x7f0c240403f0 'SIP/zvonobot-00000033' allocated [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13256] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13256] res_stasis.c: calls_0: Subscribing to 213015 [Aug 18 10:33:52] DEBUG[13256] stasis/app.c: Channel '213015' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Outgoing Call for 79821117025 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] DEBUG[13256] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13256] http.c: HTTP closing session. Top level [Aug 18 10:33:52] VERBOSE[13263] chan_sip.c: Audio is at 16466 [Aug 18 10:33:52] VERBOSE[13263] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13263] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13263] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Initializing initreq for method INVITE - callid 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117025@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK43fd6bba [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 3 [ 52]: From: ;tag=as2d218141 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 6 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13263] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117025@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK43fd6bba Max-Forwards: 70 From: ;tag=as2d218141 To: Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1233232481 1233232481 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16466 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:52] DEBUG[13263] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13264] http.c: HTTP opening session. Top level [Aug 18 10:33:52] VERBOSE[13263] dial.c: Called zvonobot/79821117025 [Aug 18 10:33:52] DEBUG[13264] http.c: HTTP Request URI is /ari/channels/213017?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117023&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13264] http.c: match request [ari/channels/213017] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13264] http.c: match request [ari/channels/213017] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13264] http.c: match request [ari/channels/213017] with handler [ari] len 3 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK43fd6bba;received=159.65.48.104 From: ;tag=as2d218141 To: ;tag=as384750c1 Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40c19384" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] DEBUG[13264] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13264] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Finding handler for channels/213017 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK43fd6bba;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d218141 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as384750c1 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40c19384" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 (Checking To) --From tag as2d218141 --To-tag as384750c1 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117025@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK43fd6bba Max-Forwards: 70 From: ;tag=as2d218141 To: ;tag=as384750c1 Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 16466 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117025@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7 Max-Forwards: 70 From: ;tag=as2d218141 To: Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117025@178.62.121.41", nonce="40c19384", response="d7e178bc39ba69fe6f6fa70821fd2f30" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1233232481 1233232482 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16466 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 From: ;tag=as2d218141 To: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d218141 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: Allocating new SIP dialog for 4c60114d666a143d56ea3a4f76cb5adc@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13260] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c20028ba0' [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) RTP allocated port 14490 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 (Checking To) --From tag as2d218141 --To-tag [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE creating session 0.0.0.0:14490 (14490) [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE create [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE add system candidates [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #7 - INVITE (got response) [Aug 18 10:33:52] DEBUG[13260] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13260] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE add candidate: 159.65.48.104:14490, 2130706431 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[13260] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13260] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE add candidate: 10.131.0.10:14490, 2130706431 [Aug 18 10:33:52] DEBUG[13260] rtp_engine.c: RTP instance '0x7f0c20028ba0' is setup and ready to go [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE stopped [Aug 18 10:33:52] DEBUG[13260] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13260] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13260] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13260] res_rtp_asterisk.c: (0x7f0c20028ba0) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13260] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13260] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: SIP call-id changed from '4c60114d666a143d56ea3a4f76cb5adc@127.0.1.1:5060' to '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13260] stasis.c: Creating topic. name: channel:213016, detail: [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13260] stasis.c: Topic 'channel:213016': 0x7f0c2002ec20 created [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Finding handler for 213017 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking channels create: Didn't match 213017 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13264] res_ari.c: Checking channels externalMedia: Didn't match 213017 [Aug 18 10:33:52] DEBUG[13264] res_ari.c: No explicit handler found for 213017. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13268] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13260] stasis.c: Creating topic. name: cache:88/channel:213016, detail: [Aug 18 10:33:52] DEBUG[13268] http.c: HTTP Request URI is /ari/channels/213020?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117020&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13268] http.c: match request [ari/channels/213020] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13268] http.c: match request [ari/channels/213020] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13268] http.c: match request [ari/channels/213020] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13268] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13268] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Finding handler for channels/213020 [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Finding handler for 213020 [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking channels create: Didn't match 213020 [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13268] res_ari.c: Checking channels externalMedia: Didn't match 213020 [Aug 18 10:33:52] DEBUG[13268] res_ari.c: No explicit handler found for 213020. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13260] stasis.c: Topic 'cache:88/channel:213016': 0x7f0c200262c0 created [Aug 18 10:33:52] DEBUG[13260] channel.c: Channel 0x7f0c20031440 'SIP/zvonobot-00000034' allocated [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13260] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13260] res_stasis.c: calls_0: Subscribing to 213016 [Aug 18 10:33:52] DEBUG[13260] stasis/app.c: Channel '213016' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Outgoing Call for 79821117024 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[13271] chan_sip.c: Audio is at 14490 [Aug 18 10:33:52] DEBUG[13260] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13260] http.c: HTTP closing session. Top level [Aug 18 10:33:52] VERBOSE[13271] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13271] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] DEBUG[13272] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13272] http.c: HTTP Request URI is /ari/channels/213018?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117022&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13272] http.c: match request [ari/channels/213018] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13272] http.c: match request [ari/channels/213018] with handler [phoneprov] len 9 [Aug 18 10:33:52] VERBOSE[13271] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13272] http.c: match request [ari/channels/213018] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13272] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13272] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Finding handler for channels/213018 [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Finding handler for 213018 [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking channels create: Didn't match 213018 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13272] res_ari.c: Checking channels externalMedia: Didn't match 213018 [Aug 18 10:33:52] DEBUG[13272] res_ari.c: No explicit handler found for 213018. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Initializing initreq for method INVITE - callid 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117024@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36aea83c [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 3 [ 52]: From: ;tag=as18b114f0 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 6 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13271] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117024@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36aea83c Max-Forwards: 70 From: ;tag=as18b114f0 To: Contact: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1189817792 1189817792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Aug 18 10:33:52] DEBUG[13271] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36aea83c;received=159.65.48.104 From: ;tag=as18b114f0 To: ;tag=as3f44aabb Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="215d54e3" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36aea83c;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as18b114f0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3f44aabb [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="215d54e3" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 (Checking To) --From tag as18b114f0 --To-tag as3f44aabb [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117024@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36aea83c Max-Forwards: 70 From: ;tag=as18b114f0 To: ;tag=as3f44aabb Contact: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 14490 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[13271] dial.c: Called zvonobot/79821117024 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117024@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33 Max-Forwards: 70 From: ;tag=as18b114f0 To: Contact: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117024@178.62.121.41", nonce="215d54e3", response="d136d346f2c40dac84737ac835ee0121" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1189817792 1189817793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13274] http.c: HTTP opening session. Top level [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 From: ;tag=as18b114f0 To: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[13274] http.c: HTTP Request URI is /ari/channels/213022?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117018&callerId=74950493843 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as18b114f0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[13274] http.c: match request [ari/channels/213022] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13274] http.c: match request [ari/channels/213022] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[13274] http.c: match request [ari/channels/213022] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13274] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13274] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Finding handler for channels/213022 [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 (Checking To) --From tag as18b114f0 --To-tag [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Finding handler for 213022 [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking channels create: Didn't match 213022 [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[13274] res_ari.c: Checking channels externalMedia: Didn't match 213022 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13274] res_ari.c: No explicit handler found for 213022. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: Allocating new SIP dialog for 1579532055e1ecd12e234dfc753d77a2@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13264] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2807d430' [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) RTP allocated port 12328 [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE creating session 0.0.0.0:12328 (12328) [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: Allocating new SIP dialog for 2217dae77fb41128687ac3a9685a2c78@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE create [Aug 18 10:33:52] DEBUG[13268] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3403efe0' [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) RTP allocated port 19268 [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE add system candidates [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE creating session 0.0.0.0:19268 (19268) [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE create [Aug 18 10:33:52] DEBUG[13264] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE add system candidates [Aug 18 10:33:52] DEBUG[13264] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13268] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE add candidate: 159.65.48.104:12328, 2130706431 [Aug 18 10:33:52] DEBUG[13268] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13264] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE add candidate: 159.65.48.104:19268, 2130706431 [Aug 18 10:33:52] DEBUG[13264] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13268] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE add candidate: 10.131.0.10:12328, 2130706431 [Aug 18 10:33:52] DEBUG[13268] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13264] rtp_engine.c: RTP instance '0x7f0c2807d430' is setup and ready to go [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) ICE stopped [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE add candidate: 10.131.0.10:19268, 2130706431 [Aug 18 10:33:52] DEBUG[13268] rtp_engine.c: RTP instance '0x7f0c3403efe0' is setup and ready to go [Aug 18 10:33:52] DEBUG[13264] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE stopped [Aug 18 10:33:52] DEBUG[13264] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13268] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13264] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13268] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13264] res_rtp_asterisk.c: (0x7f0c2807d430) RTCP setup on RTP instance [Aug 18 10:33:52] DEBUG[13268] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] VERBOSE[13264] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13275] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13268] res_rtp_asterisk.c: (0x7f0c3403efe0) RTCP setup on RTP instance [Aug 18 10:33:52] DEBUG[13264] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: SIP call-id changed from '1579532055e1ecd12e234dfc753d77a2@127.0.1.1:5060' to '1266ea01307360d60c0b6816638c526c@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13264] stasis.c: Creating topic. name: channel:213017, detail: [Aug 18 10:33:52] DEBUG[13264] stasis.c: Topic 'channel:213017': 0x7f0c2808ca00 created [Aug 18 10:33:52] DEBUG[13264] stasis.c: Creating topic. name: cache:89/channel:213017, detail: [Aug 18 10:33:52] DEBUG[13264] stasis.c: Topic 'cache:89/channel:213017': 0x7f0c2808d4e0 created [Aug 18 10:33:52] DEBUG[13275] http.c: HTTP Request URI is /ari/channels/213019?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117021&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13275] http.c: match request [ari/channels/213019] with handler [httpstatus] len 10 [Aug 18 10:33:52] VERBOSE[13268] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13275] http.c: match request [ari/channels/213019] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13268] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13275] http.c: match request [ari/channels/213019] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: SIP call-id changed from '2217dae77fb41128687ac3a9685a2c78@127.0.1.1:5060' to '636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13275] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13268] stasis.c: Creating topic. name: channel:213020, detail: [Aug 18 10:33:52] DEBUG[13275] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13268] stasis.c: Topic 'channel:213020': 0x7f0c340488e0 created [Aug 18 10:33:52] DEBUG[13268] stasis.c: Creating topic. name: cache:90/channel:213020, detail: [Aug 18 10:33:52] DEBUG[13268] stasis.c: Topic 'cache:90/channel:213020': 0x7f0c34048ce0 created [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Finding handler for channels/213019 [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13264] channel.c: Channel 0x7f0c2808a940 'SIP/zvonobot-00000035' allocated [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13264] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Finding handler for 213019 [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: Allocating new SIP dialog for 6cafdbdc2c4f9f2b2a4c826057e939b6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking channels create: Didn't match 213019 [Aug 18 10:33:52] DEBUG[13272] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c00dda0' [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) RTP allocated port 13312 [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE creating session 0.0.0.0:13312 (13312) [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE create [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE add system candidates [Aug 18 10:33:52] DEBUG[13272] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13272] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE add candidate: 159.65.48.104:13312, 2130706431 [Aug 18 10:33:52] DEBUG[13264] res_stasis.c: calls_0: Subscribing to 213017 [Aug 18 10:33:52] DEBUG[13275] res_ari.c: Checking channels externalMedia: Didn't match 213019 [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: Allocating new SIP dialog for 2d9944e86aff815b7c6895174eb7be4a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13274] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38043ba0' [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) RTP allocated port 18312 [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE creating session 0.0.0.0:18312 (18312) [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE create [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE add system candidates [Aug 18 10:33:52] DEBUG[13274] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13274] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE add candidate: 159.65.48.104:18312, 2130706431 [Aug 18 10:33:52] DEBUG[13274] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13274] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE add candidate: 10.131.0.10:18312, 2130706431 [Aug 18 10:33:52] DEBUG[13274] rtp_engine.c: RTP instance '0x7f0c38043ba0' is setup and ready to go [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE stopped [Aug 18 10:33:52] DEBUG[13274] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13274] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13274] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13274] res_rtp_asterisk.c: (0x7f0c38043ba0) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13274] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13274] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13264] stasis/app.c: Channel '213017' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13272] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13272] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE add candidate: 10.131.0.10:13312, 2130706431 [Aug 18 10:33:52] DEBUG[13264] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13275] res_ari.c: No explicit handler found for 213019. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13272] rtp_engine.c: RTP instance '0x7f0c3c00dda0' is setup and ready to go [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) ICE stopped [Aug 18 10:33:52] DEBUG[13272] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13272] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13272] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13264] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13272] res_rtp_asterisk.c: (0x7f0c3c00dda0) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13272] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13272] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: SIP call-id changed from '6cafdbdc2c4f9f2b2a4c826057e939b6@127.0.1.1:5060' to '6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13272] stasis.c: Creating topic. name: channel:213018, detail: [Aug 18 10:33:52] DEBUG[13272] stasis.c: Topic 'channel:213018': 0x7f0c3c034490 created [Aug 18 10:33:52] DEBUG[13272] stasis.c: Creating topic. name: cache:91/channel:213018, detail: [Aug 18 10:33:52] DEBUG[13272] stasis.c: Topic 'cache:91/channel:213018': 0x7f0c3c034f00 created [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: SIP call-id changed from '2d9944e86aff815b7c6895174eb7be4a@127.0.1.1:5060' to '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13277] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Outgoing Call for 79821117023 [Aug 18 10:33:52] DEBUG[13277] http.c: HTTP Request URI is /ari/channels/213021?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117019&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13274] stasis.c: Creating topic. name: channel:213022, detail: [Aug 18 10:33:52] DEBUG[13274] stasis.c: Topic 'channel:213022': 0x7f0c38048cb0 created [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2f56144c;received=159.65.48.104 From: ;tag=as396a139d To: ;tag=as46ab0e55 Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1864524172 1864524172 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 19428 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2f56144c;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as396a139d [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as46ab0e55 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1864524172 1864524172 IN IP4 178.62.121.41 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 19428 RTP/AVP 0 8 101 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 (Checking To) --From tag as396a139d --To-tag as46ab0e55 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: Allocating new SIP dialog for 790cab303ceff3a252d2da6a60c2ef81@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13275] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c4002ba40' [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) RTP allocated port 15896 [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE creating session 0.0.0.0:15896 (15896) [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE create [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE add system candidates [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13275] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13277] http.c: match request [ari/channels/213021] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13277] http.c: match request [ari/channels/213021] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:52] DEBUG[13277] http.c: match request [ari/channels/213021] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:52] DEBUG[13275] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13277] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Got SDP version 1864524172 and unique parts [root 1864524172 IN IP4 178.62.121.41] [Aug 18 10:33:52] DEBUG[13277] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1864524172 1864524172 IN IP4 178.62.121.41... OK. [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Finding handler for channels/213021 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Finding handler for 213021 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking channels create: Didn't match 213021 [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE add candidate: 159.65.48.104:15896, 2130706431 [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:52] DEBUG[13275] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:52] DEBUG[13275] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE add candidate: 10.131.0.10:15896, 2130706431 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:52] DEBUG[13275] rtp_engine.c: RTP instance '0x7f0c4002ba40' is setup and ready to go [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) ICE stopped [Aug 18 10:33:52] DEBUG[13277] res_ari.c: Checking channels externalMedia: Didn't match 213021 [Aug 18 10:33:52] DEBUG[13277] res_ari.c: No explicit handler found for 213021. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13275] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE set role failed; no ice instance [Aug 18 10:33:52] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13274] stasis.c: Creating topic. name: cache:92/channel:213022, detail: [Aug 18 10:33:52] DEBUG[13275] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTCP setting address on RTP instance [Aug 18 10:33:52] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c8001de30 -- Strict RTP learning after remote address set to: 178.62.121.41:19428 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:19428 [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0086e88) from 0x7f0c147e2330 to 0x7f0c8001c8c8 [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0052878) from 0x7f0c147e2330 to 0x7f0c8001c8c8 [Aug 18 10:33:52] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0089b08) from 0x7f0c147e2330 to 0x7f0c8001c8c8 [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTCP ignoring duplicate property [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:52] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001b setting read format path: alaw -> alaw [Aug 18 10:33:52] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001b setting write format path: alaw -> alaw [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS - ast_rtp_activate rtp=0x7f0c8001de30 - setup and perform DTLS' [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001de30) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:52] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001de30) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:52] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:52] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Strict routing enforced for session 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:52] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:52] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117047@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79061431 Max-Forwards: 70 From: ;tag=as396a139d To: ;tag=as46ab0e55 Contact: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[13044] dial.c: SIP/zvonobot-0000001b answered [Aug 18 10:33:52] VERBOSE[13044] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000001b [Aug 18 10:33:52] DEBUG[13044] stasis/app.c: Channel '212993' is 2 interested in calls_0 [Aug 18 10:33:52] DEBUG[13275] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13275] res_rtp_asterisk.c: (0x7f0c4002ba40) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13275] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Session timer started: 9 - 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 1768000ms [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:52] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:52] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:52] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13279] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13274] stasis.c: Topic 'cache:92/channel:213022': 0x7f0c38048e90 created [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13278] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] DEBUG[13275] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] VERBOSE[13276] chan_sip.c: Audio is at 12328 [Aug 18 10:33:52] VERBOSE[13276] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] DEBUG[13279] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: SIP call-id changed from '790cab303ceff3a252d2da6a60c2ef81@127.0.1.1:5060' to '4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13279] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:52] VERBOSE[13276] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] DEBUG[13278] http.c: HTTP Request URI is /ari/channels/213023?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117017&callerId=74950493843 [Aug 18 10:33:52] DEBUG[13279] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13275] stasis.c: Creating topic. name: channel:213019, detail: [Aug 18 10:33:52] DEBUG[13268] channel.c: Channel 0x7f0c34047590 'SIP/zvonobot-00000036' allocated [Aug 18 10:33:52] DEBUG[13279] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13279] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13275] stasis.c: Topic 'channel:213019': 0x7f0c40036cb0 created [Aug 18 10:33:52] DEBUG[13275] stasis.c: Creating topic. name: cache:93/channel:213019, detail: [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] VERBOSE[13276] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13275] stasis.c: Topic 'cache:93/channel:213019': 0x7f0c40036250 created [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13278] http.c: match request [ari/channels/213023] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13268] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Initializing initreq for method INVITE - callid 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13278] http.c: match request [ari/channels/213023] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117023@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] DEBUG[13278] http.c: match request [ari/channels/213023] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13278] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13279] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13279] stasis.c: Creating topic. name: bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59, detail: [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: Allocating new SIP dialog for 33834788197e5efb6939e9f75800b000@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[13278] http.c: HTTP consuming request body [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Finding handler for channels/213023 [Aug 18 10:33:52] DEBUG[13279] stasis.c: Topic 'bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': 0x7f0c78007260 created [Aug 18 10:33:52] DEBUG[13277] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c70049e60' [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) RTP allocated port 14346 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5431df89 [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE creating session 0.0.0.0:14346 (14346) [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE create [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 3 [ 52]: From: ;tag=as2f1904c0 [Aug 18 10:33:52] DEBUG[13279] stasis.c: Creating topic. name: cache:94/bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59, detail: [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 6 [ 60]: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE add system candidates [Aug 18 10:33:52] DEBUG[13268] res_stasis.c: calls_0: Subscribing to 213020 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13279] stasis.c: Topic 'cache:94/bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': 0x7f0c78017da0 created [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13277] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13268] stasis/app.c: Channel '213020' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13277] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13268] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13279] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as two channels are required [Aug 18 10:33:52] DEBUG[13279] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13279] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13279] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:52] DEBUG[13279] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13268] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13279] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology constructor [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Outgoing Call for 79821117020 [Aug 18 10:33:52] DEBUG[13279] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology start [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Finding handler for 213023 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking channels create: Didn't match 213023 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] VERBOSE[13276] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117023@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5431df89 Max-Forwards: 70 From: ;tag=as2f1904c0 To: Contact: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2004171457 2004171457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12328 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE add candidate: 159.65.48.104:14346, 2130706431 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:52] DEBUG[13276] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13278] res_ari.c: Checking channels externalMedia: Didn't match 213023 [Aug 18 10:33:52] DEBUG[13279] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13278] res_ari.c: No explicit handler found for 213023. Using wildcard channelId. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5431df89;received=159.65.48.104 From: ;tag=as2f1904c0 To: ;tag=as274f7c2e Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46800fce" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5431df89;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2f1904c0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as274f7c2e [Aug 18 10:33:52] DEBUG[13279] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13277] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46800fce" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 (Checking To) --From tag as2f1904c0 --To-tag as274f7c2e [Aug 18 10:33:52] DEBUG[13277] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13281] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE add candidate: 10.131.0.10:14346, 2130706431 [Aug 18 10:33:52] DEBUG[13281] http.c: HTTP Request URI is /ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/addChannel?channel=212993 [Aug 18 10:33:52] DEBUG[13277] rtp_engine.c: RTP instance '0x7f0c70049e60' is setup and ready to go [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '1266ea01307360d60c0b6816638c526c@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117023@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5431df89 Max-Forwards: 70 From: ;tag=as2f1904c0 To: ;tag=as274f7c2e Contact: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 12328 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) ICE stopped [Aug 18 10:33:52] DEBUG[13277] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13277] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] DEBUG[13277] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13277] res_rtp_asterisk.c: (0x7f0c70049e60) RTCP setup on RTP instance [Aug 18 10:33:52] VERBOSE[13277] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[13277] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: SIP call-id changed from '33834788197e5efb6939e9f75800b000@127.0.1.1:5060' to '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[13277] stasis.c: Creating topic. name: channel:213021, detail: [Aug 18 10:33:52] DEBUG[13277] stasis.c: Topic 'channel:213021': 0x7f0c7005b360 created [Aug 18 10:33:52] DEBUG[13277] stasis.c: Creating topic. name: cache:95/channel:213021, detail: [Aug 18 10:33:52] DEBUG[13277] stasis.c: Topic 'cache:95/channel:213021': 0x7f0c7005bdd0 created [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117023@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36349d97 Max-Forwards: 70 From: ;tag=as2f1904c0 To: Contact: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117023@178.62.121.41", nonce="46800fce", response="4d6a3e76b7b71cc92aa47ea72d3b8cce" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2004171457 2004171458 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12328 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:52] DEBUG[13281] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13281] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13281] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/addChannel] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13281] http.c: Match made with [ari] [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36349d97;received=159.65.48.104 From: ;tag=as2f1904c0 To: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Finding handler for bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/addChannel [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36349d97;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2f1904c0 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Finding handler for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13281] res_ari.c: No explicit handler found for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59. Using wildcard bridgeId. [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Finding handler for addChannel [Aug 18 10:33:52] DEBUG[13281] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:52] DEBUG[13281] stasis/control.c: 212993: Sending channel add_to_bridge command [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1266ea01307360d60c0b6816638c526c@159.65.48.104:5060 (Checking To) --From tag as2f1904c0 --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #6 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1266ea01307360d60c0b6816638c526c@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] VERBOSE[13276] dial.c: Called zvonobot/79821117023 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[13280] chan_sip.c: Audio is at 19268 [Aug 18 10:33:52] VERBOSE[13280] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13280] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13280] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Initializing initreq for method INVITE - callid 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117020@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a4bba6e [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 3 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 6 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13280] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117020@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a4bba6e Max-Forwards: 70 From: ;tag=as366f0ed0 To: Contact: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1162804906 1162804906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19268 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:52] DEBUG[13280] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a4bba6e;received=159.65.48.104 From: ;tag=as366f0ed0 To: ;tag=as196ece37 Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f8ac59d" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a4bba6e;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as196ece37 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f8ac59d" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 (Checking To) --From tag as366f0ed0 --To-tag as196ece37 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117020@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a4bba6e Max-Forwards: 70 From: ;tag=as366f0ed0 To: ;tag=as196ece37 Contact: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 19268 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117020@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b Max-Forwards: 70 From: ;tag=as366f0ed0 To: Contact: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117020@178.62.121.41", nonce="3f8ac59d", response="1e8a015b9a87fc1fb3fc135d67e947df" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1162804906 1162804907 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19268 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 From: ;tag=as366f0ed0 To: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP got report of 100 bytes from 178.62.121.41:11671 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[13272] channel.c: Channel 0x7f0c3c032360 'SIP/zvonobot-00000037' allocated [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13272] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 (Checking To) --From tag as366f0ed0 --To-tag [Aug 18 10:33:52] DEBUG[13272] res_stasis.c: calls_0: Subscribing to 213018 [Aug 18 10:33:52] DEBUG[13272] stasis/app.c: Channel '213018' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13272] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13272] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #6 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Outgoing Call for 79821117022 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[13283] chan_sip.c: Audio is at 13312 [Aug 18 10:33:52] VERBOSE[13283] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13283] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13283] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Initializing initreq for method INVITE - callid 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117022@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220b5686 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 3 [ 52]: From: ;tag=as17b5614f [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 6 [ 60]: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13283] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117022@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220b5686 Max-Forwards: 70 From: ;tag=as17b5614f To: Contact: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1199100009 1199100009 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13312 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:52] DEBUG[13283] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[13280] dial.c: Called zvonobot/79821117020 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[13283] dial.c: Called zvonobot/79821117022 [Aug 18 10:33:52] DEBUG[13274] channel.c: Channel 0x7f0c3804f760 'SIP/zvonobot-00000038' allocated [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13274] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[13274] res_stasis.c: calls_0: Subscribing to 213022 [Aug 18 10:33:52] DEBUG[13274] stasis/app.c: Channel '213022' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13274] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13275] channel.c: Channel 0x7f0c40036ff0 'SIP/zvonobot-00000039' allocated [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220b5686;received=159.65.48.104 From: ;tag=as17b5614f To: ;tag=as224c0fd3 Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fc5eb61" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220b5686;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as17b5614f [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as224c0fd3 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13274] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13275] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fc5eb61" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Outgoing Call for 79821117018 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 (Checking To) --From tag as17b5614f --To-tag as224c0fd3 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117022@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220b5686 Max-Forwards: 70 From: ;tag=as17b5614f To: ;tag=as224c0fd3 Contact: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 13312 [Aug 18 10:33:52] VERBOSE[13286] chan_sip.c: Audio is at 18312 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13286] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] DEBUG[13275] res_stasis.c: calls_0: Subscribing to 213019 [Aug 18 10:33:52] VERBOSE[13286] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13275] stasis/app.c: Channel '213019' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] VERBOSE[13286] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117022@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30384a82 Max-Forwards: 70 From: ;tag=as17b5614f To: Contact: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117022@178.62.121.41", nonce="2fc5eb61", response="d07385c0cdd52dce7e1be671dc6613f5" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1199100009 1199100010 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13312 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13275] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Outgoing Call for 79821117021 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Initializing initreq for method INVITE - callid 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13275] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117018@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ce564ac [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 3 [ 52]: From: ;tag=as009e460a [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 6 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30384a82;received=159.65.48.104 From: ;tag=as17b5614f To: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30384a82;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as17b5614f [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] VERBOSE[13286] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117018@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ce564ac Max-Forwards: 70 From: ;tag=as009e460a To: Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1666136377 1666136377 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18312 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13286] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: Allocating new SIP dialog for 45189b286057f9510a55587b5ee26c4b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060 (Checking To) --From tag as17b5614f --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #6 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6c91598655a5f0a7462d85a451a428b4@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13278] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c01e650' [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP allocated port 11012 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] VERBOSE[13286] dial.c: Called zvonobot/79821117018 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[13287] chan_sip.c: Audio is at 15896 [Aug 18 10:33:52] VERBOSE[13287] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13287] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13287] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13277] channel.c: Channel 0x7f0c700595e0 'SIP/zvonobot-0000003a' allocated [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE creating session 0.0.0.0:11012 (11012) [Aug 18 10:33:52] DEBUG[13277] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ce564ac;received=159.65.48.104 From: ;tag=as009e460a To: ;tag=as1bc2ee5e Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68db3dac" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ce564ac;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as009e460a [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1bc2ee5e [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68db3dac" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE create [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 (Checking To) --From tag as009e460a --To-tag as1bc2ee5e [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Initializing initreq for method INVITE - callid 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117021@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3aa8ae6c [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 3 [ 52]: From: ;tag=as601f237f [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117018@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ce564ac Max-Forwards: 70 From: ;tag=as009e460a To: ;tag=as1bc2ee5e Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13277] res_stasis.c: calls_0: Subscribing to 213021 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 18312 [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 6 [ 60]: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13277] stasis/app.c: Channel '213021' is 1 interested in calls_0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] DEBUG[13277] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13277] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Outgoing Call for 79821117019 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117018@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac Max-Forwards: 70 From: ;tag=as009e460a To: Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117018@178.62.121.41", nonce="68db3dac", response="566179458fac1e2df89f61c0e62320a4" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1666136377 1666136378 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18312 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE add system candidates [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13278] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] VERBOSE[13289] chan_sip.c: Audio is at 14346 [Aug 18 10:33:52] DEBUG[13278] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] VERBOSE[13289] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13289] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE add candidate: 159.65.48.104:11012, 2130706431 [Aug 18 10:33:52] DEBUG[13278] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] VERBOSE[13289] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13278] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE add candidate: 10.131.0.10:11012, 2130706431 [Aug 18 10:33:52] DEBUG[13278] rtp_engine.c: RTP instance '0x7f0c7c01e650' is setup and ready to go [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE stopped [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13278] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13287] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117021@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3aa8ae6c Max-Forwards: 70 From: ;tag=as601f237f To: Contact: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1704128100 1704128100 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15896 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13278] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 From: ;tag=as009e460a To: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as009e460a [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Initializing initreq for method INVITE - callid 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117019@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cc16fad [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 3 [ 52]: From: ;tag=as5e4952fc [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13278] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 6 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[13278] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP setup on RTP instance [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 (Checking To) --From tag as009e460a --To-tag [Aug 18 10:33:52] VERBOSE[13278] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #6 - INVITE (got response) [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] VERBOSE[13289] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117019@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cc16fad Max-Forwards: 70 From: ;tag=as5e4952fc To: Contact: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 379999143 379999143 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14346 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13278] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:52] DEBUG[13287] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Aug 18 10:33:52] DEBUG[13289] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:52] VERBOSE[13287] dial.c: Called zvonobot/79821117021 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3aa8ae6c;received=159.65.48.104 From: ;tag=as601f237f To: ;tag=as604423bc Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="198e8d54" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3aa8ae6c;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: SIP call-id changed from '45189b286057f9510a55587b5ee26c4b@127.0.1.1:5060' to '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as601f237f [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[13289] dial.c: Called zvonobot/79821117019 [Aug 18 10:33:52] DEBUG[13278] stasis.c: Creating topic. name: channel:213023, detail: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as604423bc [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[13278] stasis.c: Topic 'channel:213023': 0x7f0c7c03df90 created [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13278] stasis.c: Creating topic. name: cache:96/channel:213023, detail: [Aug 18 10:33:52] DEBUG[13278] stasis.c: Topic 'cache:96/channel:213023': 0x7f0c7c03f5a0 created [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="198e8d54" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 (Checking To) --From tag as601f237f --To-tag as604423bc [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117021@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3aa8ae6c Max-Forwards: 70 From: ;tag=as601f237f To: ;tag=as604423bc Contact: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 15896 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117021@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709 Max-Forwards: 70 From: ;tag=as601f237f To: Contact: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117021@178.62.121.41", nonce="198e8d54", response="cd1d5bc2e39434ca4b97925133f003ca" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1704128100 1704128101 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15896 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cc16fad;received=159.65.48.104 From: ;tag=as5e4952fc To: ;tag=as035e3a49 Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49e49fba" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cc16fad;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5e4952fc [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as035e3a49 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49e49fba" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 (Checking To) --From tag as5e4952fc --To-tag as035e3a49 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117019@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cc16fad Max-Forwards: 70 From: ;tag=as5e4952fc To: ;tag=as035e3a49 Contact: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 14346 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117019@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647 Max-Forwards: 70 From: ;tag=as5e4952fc To: Contact: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117019@178.62.121.41", nonce="49e49fba", response="3155c6295b0149287f721b5bf5a9b25a" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 379999143 379999144 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14346 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 From: ;tag=as601f237f To: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as601f237f [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 (Checking To) --From tag as601f237f --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #11 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 From: ;tag=as5e4952fc To: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5e4952fc [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13278] channel.c: Channel 0x7f0c7c03c750 'SIP/zvonobot-0000003b' allocated [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:52] DEBUG[13278] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 (Checking To) --From tag as5e4952fc --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13278] res_stasis.c: calls_0: Subscribing to 213023 [Aug 18 10:33:52] DEBUG[13278] stasis/app.c: Channel '213023' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Outgoing Call for 79821117017 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:52] DEBUG[13278] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13278] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[13292] chan_sip.c: Audio is at 11012 [Aug 18 10:33:52] VERBOSE[13292] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[13292] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[13292] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Initializing initreq for method INVITE - callid 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117017@178.62.121.41 SIP/2.0 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c92b7fa [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 3 [ 52]: From: ;tag=as4f7b8e6e [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 6 [ 60]: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:52 GMT [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:52] VERBOSE[13292] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117017@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c92b7fa Max-Forwards: 70 From: ;tag=as4f7b8e6e To: Contact: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1727194744 1727194744 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11012 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:33:52] DEBUG[13292] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:52] VERBOSE[13292] dial.c: Called zvonobot/79821117017 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c92b7fa;received=159.65.48.104 From: ;tag=as4f7b8e6e To: ;tag=as1cfe9620 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0de4ceb7" Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c92b7fa;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4f7b8e6e [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1cfe9620 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0de4ceb7" [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 (Checking To) --From tag as4f7b8e6e --To-tag as1cfe9620 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Stopping retransmission on '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117017@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c92b7fa Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as1cfe9620 Contact: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Audio is at 11012 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117017@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e Max-Forwards: 70 From: ;tag=as4f7b8e6e To: Contact: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41", nonce="0de4ceb7", response="1dc513df18b744e673de6d5bf91db9b6" Date: Wed, 18 Aug 2021 10:33:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1727194744 1727194745 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11012 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 From: ;tag=as4f7b8e6e To: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4f7b8e6e [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:52] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 (Checking To) --From tag as4f7b8e6e --To-tag [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8 - INVITE (got response) [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:52] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:52] DEBUG[13044] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000001b [Aug 18 10:33:52] DEBUG[13044] stasis/control.c: 212993: Adding to bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:33:52] DEBUG[13044] stasis/app.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13294] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is joining [Aug 18 10:33:52] DEBUG[13294] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pushing 0x7f0c8c00b190(SIP/zvonobot-0000001b) [Aug 18 10:33:52] VERBOSE[13294] bridge_channel.c: Channel SIP/zvonobot-0000001b joined 'simple_bridge' stasis-bridge [Aug 18 10:33:52] DEBUG[13294] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as two channels are required [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 is already using the new technology. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13294] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTP changing ssrc from 1478669804 to 1496765868 due to a source change [Aug 18 10:33:52] DEBUG[13044] stasis/app.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' is 2 interested in calls_0 [Aug 18 10:33:52] DEBUG[13281] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:52] DEBUG[13281] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13296] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13296] http.c: HTTP Request URI is /ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/record?name=212993_qshPOCtSgcnLuvCpzOgPjMQITTkRHVMS&format=wav [Aug 18 10:33:52] DEBUG[13296] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/record] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13296] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/record] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13296] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/record] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13296] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Finding handler for bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Finding handler for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13296] res_ari.c: No explicit handler found for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59. Using wildcard bridgeId. [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Finding handler for record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:52] DEBUG[13296] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:52] DEBUG[13296] stasis.c: Creating topic. name: channel:1629282832.83, detail: [Aug 18 10:33:52] DEBUG[13296] stasis.c: Topic 'channel:1629282832.83': 0x7f0c98034880 created [Aug 18 10:33:52] DEBUG[13296] stasis.c: Creating topic. name: cache:97/channel:1629282832.83, detail: [Aug 18 10:33:52] DEBUG[13296] stasis.c: Topic 'cache:97/channel:1629282832.83': 0x7f0c98022fd0 created [Aug 18 10:33:52] DEBUG[13296] channel.c: Channel 0x7f0c9803f2e0 'Recorder/ARI-00000007;1' allocated [Aug 18 10:33:52] DEBUG[13296] stasis.c: Creating topic. name: channel:1629282832.84, detail: [Aug 18 10:33:52] DEBUG[13296] stasis.c: Topic 'channel:1629282832.84': 0x7f0c9802db40 created [Aug 18 10:33:52] DEBUG[13296] stasis.c: Creating topic. name: cache:98/channel:1629282832.84, detail: [Aug 18 10:33:52] DEBUG[13296] stasis.c: Topic 'cache:98/channel:1629282832.84': 0x7f0c9803ccc0 created [Aug 18 10:33:52] DEBUG[13296] channel.c: Channel 0x7f0c980450a0 'Recorder/ARI-00000007;2' allocated [Aug 18 10:33:52] DEBUG[13296] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:52] DEBUG[13298] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining [Aug 18 10:33:52] DEBUG[13298] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pushing 0x7f0c9802d570(Recorder/ARI-00000007;2) [Aug 18 10:33:52] DEBUG[13298] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:52] VERBOSE[13298] bridge_channel.c: Channel Recorder/ARI-00000007;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:52] DEBUG[13298] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59'. Checking compatability for channels 'SIP/zvonobot-0000001b' and 'Recorder/ARI-00000007;2' [Aug 18 10:33:52] DEBUG[13298] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as could not get details [Aug 18 10:33:52] DEBUG[13298] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13298] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13298] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13298] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13298] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 is already using the new technology. [Aug 18 10:33:52] DEBUG[13298] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13298] channel.c: Channel Recorder/ARI-00000007;2 setting read format path: slin -> slin [Aug 18 10:33:52] DEBUG[13298] channel.c: Channel SIP/zvonobot-0000001b setting write format path: slin -> alaw [Aug 18 10:33:52] DEBUG[13298] channel.c: Channel SIP/zvonobot-0000001b setting read format path: alaw -> slin [Aug 18 10:33:52] DEBUG[13298] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13296] res_stasis_recording.c: 1629282832.83: Sending record(212993_qshPOCtSgcnLuvCpzOgPjMQITTkRHVMS.wav) command [Aug 18 10:33:52] DEBUG[13296] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:52] DEBUG[13296] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13300] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13299] app.c: play_and_record: , /var/spool/asterisk/recording/212993_qshPOCtSgcnLuvCpzOgPjMQITTkRHVMS, 'wav' [Aug 18 10:33:52] DEBUG[13300] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:52] DEBUG[13300] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13300] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13300] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13300] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13299] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] VERBOSE[13299] app.c: x=0, open writing: /var/spool/asterisk/recording/212993_qshPOCtSgcnLuvCpzOgPjMQITTkRHVMS format: wav, 0x7f0ca40610c0 [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13300] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13300] stasis.c: Creating topic. name: bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2, detail: [Aug 18 10:33:52] DEBUG[13300] stasis.c: Topic 'bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2': 0x7f0cb007f410 created [Aug 18 10:33:52] DEBUG[13300] stasis.c: Creating topic. name: cache:99/bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2, detail: [Aug 18 10:33:52] DEBUG[13300] stasis.c: Topic 'cache:99/bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2': 0x7f0cb008d500 created [Aug 18 10:33:52] DEBUG[13300] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:52] DEBUG[13300] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13300] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13300] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:52] DEBUG[13300] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13300] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: calling simple_bridge technology constructor [Aug 18 10:33:52] DEBUG[13300] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: calling simple_bridge technology start [Aug 18 10:33:52] DEBUG[13300] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13300] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13301] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13301] http.c: HTTP Request URI is /ari/channels/212993/snoop?app=calls_0&spy=in [Aug 18 10:33:52] DEBUG[13301] http.c: match request [ari/channels/212993/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13301] http.c: match request [ari/channels/212993/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13301] http.c: match request [ari/channels/212993/snoop] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13301] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Finding handler for channels/212993/snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Finding handler for 212993 [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channels create: Didn't match 212993 [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channels externalMedia: Didn't match 212993 [Aug 18 10:33:52] DEBUG[13301] res_ari.c: No explicit handler found for 212993. Using wildcard channelId. [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Finding handler for snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:52] DEBUG[13301] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:52] DEBUG[13301] stasis.c: Creating topic. name: channel:1629282832.85, detail: [Aug 18 10:33:52] DEBUG[13301] stasis.c: Topic 'channel:1629282832.85': 0x7f0cac01e180 created [Aug 18 10:33:52] DEBUG[13301] stasis.c: Creating topic. name: cache:100/channel:1629282832.85, detail: [Aug 18 10:33:52] DEBUG[13301] stasis.c: Topic 'cache:100/channel:1629282832.85': 0x7f0cac01cfb0 created [Aug 18 10:33:52] DEBUG[13301] channel.c: Channel 0x7f0cac03e430 'Snoop/212993-00000003' allocated [Aug 18 10:33:52] DEBUG[13294] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59'. Checking compatability for channels 'SIP/zvonobot-0000001b' and 'Recorder/ARI-00000007;2' [Aug 18 10:33:52] DEBUG[13294] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as channel 'SIP/zvonobot-0000001b' has features which prevent it [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13294] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 is already using the new technology. [Aug 18 10:33:52] DEBUG[13301] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13306] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13302] stasis/app.c: Channel '1629282832.85' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13306] http.c: HTTP Request URI is /ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play?media=sound%3Asilence%2F2 [Aug 18 10:33:52] DEBUG[13306] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13306] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13306] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13306] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Finding handler for bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Finding handler for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13306] res_ari.c: No explicit handler found for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59. Using wildcard bridgeId. [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Finding handler for play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:52] DEBUG[13306] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:52] DEBUG[13306] stasis.c: Creating topic. name: channel:1629282832.86, detail: [Aug 18 10:33:52] DEBUG[13306] stasis.c: Topic 'channel:1629282832.86': 0x2c35110 created [Aug 18 10:33:52] DEBUG[13306] stasis.c: Creating topic. name: cache:101/channel:1629282832.86, detail: [Aug 18 10:33:52] DEBUG[13306] stasis.c: Topic 'cache:101/channel:1629282832.86': 0x2c35b40 created [Aug 18 10:33:52] DEBUG[13306] channel.c: Channel 0x2c32fc0 'Announcer/ARI-00000008;1' allocated [Aug 18 10:33:52] DEBUG[13306] stasis.c: Creating topic. name: channel:1629282832.87, detail: [Aug 18 10:33:52] DEBUG[13306] stasis.c: Topic 'channel:1629282832.87': 0x2c3d590 created [Aug 18 10:33:52] DEBUG[13306] stasis.c: Creating topic. name: cache:102/channel:1629282832.87, detail: [Aug 18 10:33:52] DEBUG[13306] stasis.c: Topic 'cache:102/channel:1629282832.87': 0x2c3e010 created [Aug 18 10:33:52] DEBUG[13301] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13306] channel.c: Channel 0x2c3b150 'Announcer/ARI-00000008;2' allocated [Aug 18 10:33:52] DEBUG[13306] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 457 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 457 [Aug 18 10:33:52] DEBUG[13306] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000008;1' [Aug 18 10:33:52] DEBUG[13308] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13309] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x2c3d160(Announcer/ARI-00000008;2) is joining [Aug 18 10:33:52] DEBUG[13308] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212993&app=calls_0&format=slin16&external_host=127.0.0.1%3A50409 [Aug 18 10:33:52] DEBUG[13309] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pushing 0x2c3d160(Announcer/ARI-00000008;2) [Aug 18 10:33:52] DEBUG[13308] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13308] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13308] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13308] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Finding handler for channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:52] DEBUG[13309] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:52] VERBOSE[13309] bridge_channel.c: Channel Announcer/ARI-00000008;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13308] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13309] bridge.c: Chose bridge technology softmix [Aug 18 10:33:52] VERBOSE[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: switching from simple_bridge technology to softmix [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology constructor [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c8c00b190(SIP/zvonobot-0000001b) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c9802d570(Recorder/ARI-00000007;2) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is leaving simple_bridge technology (dummy) [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology stop [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x2c3d160(Announcer/ARI-00000008;2) is joining softmix technology [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Announcer/ARI-00000008;2: [Aug 18 10:33:52] DEBUG[13309] channel.c: Channel Announcer/ARI-00000008;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13308] netsock2.c: Splitting '127.0.0.1:50409' into... [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Announcer/ARI-00000008;2: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Announcer/ARI-00000008;2: Not in SFU mode [Aug 18 10:33:52] DEBUG[13308] netsock2.c: ...host '127.0.0.1' and port '50409'. [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is joining softmix technology [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: SIP/zvonobot-0000001b: [Aug 18 10:33:52] DEBUG[13308] netsock2.c: Splitting '127.0.0.1:50409' into... [Aug 18 10:33:52] DEBUG[13308] netsock2.c: ...host '127.0.0.1' and port '50409'. [Aug 18 10:33:52] DEBUG[13308] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:52] DEBUG[13308] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c10045b20' [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) RTP allocated port 15898 [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) ICE creating session 127.0.0.1:15898 (15898) [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: SIP/zvonobot-0000001b: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: SIP/zvonobot-0000001b: Not in SFU mode [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining softmix technology [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Recorder/ARI-00000007;2: [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) ICE create [Aug 18 10:33:52] DEBUG[13309] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13309] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Recorder/ARI-00000007;2: [Aug 18 10:33:52] DEBUG[13309] bridge_softmix.c: Recorder/ARI-00000007;2: Not in SFU mode [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology start [Aug 18 10:33:52] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology destructor [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) ICE add system candidates [Aug 18 10:33:52] DEBUG[13308] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:52] DEBUG[13308] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) ICE add candidate: 159.65.48.104:15898, 2130706431 [Aug 18 10:33:52] DEBUG[13308] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:52] DEBUG[13310] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: starting mixing thread [Aug 18 10:33:52] DEBUG[13306] res_stasis_playback.c: 1629282832.86: Sending play(sound:silence/2) command [Aug 18 10:33:52] DEBUG[13294] audiohook.c: Audiohook 0x7f0cac03acc0 has stale audio in its factories. Flushing them both [Aug 18 10:33:52] DEBUG[13294] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTP ooh, format changed from none to alaw [Aug 18 10:33:52] DEBUG[13294] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTCP starting transmission [Aug 18 10:33:52] DEBUG[13306] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:52] VERBOSE[13294] res_rtp_asterisk.c: 0x7f0c8001de30 -- Strict RTP switching to RTP target address 178.62.121.41:19428 as source [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:52] DEBUG[13294] audiohook.c: Audiohook 0x7f0cac03acc0 has stale audio in its factories. Flushing them both [Aug 18 10:33:52] DEBUG[13311] channel.c: Channel Announcer/ARI-00000008;1 setting write format path: gsm -> slin [Aug 18 10:33:52] DEBUG[13306] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:52] DEBUG[13308] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:52] DEBUG[13308] res_rtp_asterisk.c: (0x7f0c10045b20) ICE add candidate: 10.131.0.10:15898, 2130706431 [Aug 18 10:33:52] DEBUG[13308] rtp_engine.c: RTP instance '0x7f0c10045b20' is setup and ready to go [Aug 18 10:33:52] DEBUG[13308] stasis.c: Creating topic. name: channel:robot_212993, detail: [Aug 18 10:33:52] DEBUG[13308] stasis.c: Topic 'channel:robot_212993': 0x7f0c1004f9f0 created [Aug 18 10:33:52] DEBUG[13308] stasis.c: Creating topic. name: cache:103/channel:robot_212993, detail: [Aug 18 10:33:52] DEBUG[13308] stasis.c: Topic 'cache:103/channel:robot_212993': 0x7f0c1004fc00 created [Aug 18 10:33:52] DEBUG[13311] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:52] VERBOSE[13311] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:52] DEBUG[13308] channel.c: Channel 0x7f0c1004e480 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20' allocated [Aug 18 10:33:52] DEBUG[13308] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:52] VERBOSE[13308] res_rtp_asterisk.c: 0x7f0c10047270 -- Strict RTP learning after remote address set to: 127.0.0.1:50409 [Aug 18 10:33:52] DEBUG[13308] res_stasis.c: calls_0: Subscribing to robot_212993 [Aug 18 10:33:52] DEBUG[13308] stasis/app.c: Channel 'robot_212993' is 1 interested in calls_0 [Aug 18 10:33:52] VERBOSE[13312] dial.c: Called 127.0.0.1:50409 [Aug 18 10:33:52] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50409 [Aug 18 10:33:52] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50409 - state 2 (In use) [Aug 18 10:33:52] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50409, detail: [Aug 18 10:33:52] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50409': 0x7f0c84052db0 created [Aug 18 10:33:52] VERBOSE[13312] dial.c: UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 answered [Aug 18 10:33:52] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50409' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:52] VERBOSE[13312] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:52] DEBUG[13308] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:52] DEBUG[13308] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13312] stasis/app.c: Channel 'robot_212993' is 2 interested in calls_0 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:52] DEBUG[13313] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13313] http.c: HTTP Request URI is /ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2/addChannel?channel=1629282832.85%2Crobot_212993 [Aug 18 10:33:52] DEBUG[13313] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13313] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13313] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2/addChannel] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13313] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Finding handler for bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2/addChannel [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Finding handler for 9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13313] res_ari.c: No explicit handler found for 9d1bf1e2-893f-4249-b006-4b3a345e76a2. Using wildcard bridgeId. [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Finding handler for addChannel [Aug 18 10:33:52] DEBUG[13313] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:52] DEBUG[13313] stasis/control.c: 1629282832.85: Sending channel add_to_bridge command [Aug 18 10:33:52] DEBUG[13302] bridge_roles.c: Roles did not exist on channel Snoop/212993-00000003 [Aug 18 10:33:52] DEBUG[13302] stasis/control.c: 1629282832.85: Adding to bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:33:52] DEBUG[13302] stasis/app.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' is 1 interested in calls_0 [Aug 18 10:33:52] DEBUG[13314] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0cb4036220(Snoop/212993-00000003) is joining [Aug 18 10:33:52] DEBUG[13314] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: pushing 0x7f0cb4036220(Snoop/212993-00000003) [Aug 18 10:33:52] VERBOSE[13314] bridge_channel.c: Channel Snoop/212993-00000003 joined 'simple_bridge' stasis-bridge <9d1bf1e2-893f-4249-b006-4b3a345e76a2> [Aug 18 10:33:52] DEBUG[13314] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:52] DEBUG[13314] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13314] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13314] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13314] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13314] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 is already using the new technology. [Aug 18 10:33:52] DEBUG[13314] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0cb4036220(Snoop/212993-00000003) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13313] stasis/control.c: robot_212993: Sending channel add_to_bridge command [Aug 18 10:33:52] DEBUG[13302] stasis/app.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' is 2 interested in calls_0 [Aug 18 10:33:52] DEBUG[13224] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:52] DEBUG[13224] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:52] DEBUG[13224] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:52] DEBUG[13224] channel.c: Channel Announcer/ARI-00000004;1 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13224] channel.c: Channel 0x7f0c2c00b5e0 'Announcer/ARI-00000004;1' hanging up. Refs: 2 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:52] DEBUG[13315] http.c: HTTP opening session. Top level [Aug 18 10:33:52] DEBUG[13224] channel.c: Channel 0x7f0c2c00b5e0 'Announcer/ARI-00000004;1' destroying [Aug 18 10:33:52] DEBUG[13222] bridge_channel.c: Setting 0x7f0c2c00ad50(Announcer/ARI-00000004;2) state from:0 to:1 [Aug 18 10:33:52] DEBUG[13315] http.c: HTTP Request URI is /ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:52] DEBUG[13315] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [httpstatus] len 10 [Aug 18 10:33:52] DEBUG[13315] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [phoneprov] len 9 [Aug 18 10:33:52] DEBUG[13315] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play] with handler [ari] len 3 [Aug 18 10:33:52] DEBUG[13222] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pulling 0x7f0c2c00ad50(Announcer/ARI-00000004;2) [Aug 18 10:33:52] DEBUG[13315] http.c: Match made with [ari] [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Finding handler for bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6/play [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Finding handler for bridges [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:52] VERBOSE[13222] bridge_channel.c: Channel Announcer/ARI-00000004;2 left 'softmix' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:33:52] DEBUG[13222] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2c00ad50(Announcer/ARI-00000004;2) is leaving softmix technology [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:52] DEBUG[13222] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6'. Checking compatability for channels 'SIP/zvonobot-00000012' and 'Recorder/ARI-00000003;2' [Aug 18 10:33:52] DEBUG[13222] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as channel 'SIP/zvonobot-00000012' has features which prevent it [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Finding handler for 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13222] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] VERBOSE[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: switching from softmix technology to simple_bridge [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology constructor [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c080248a0(SIP/zvonobot-00000012) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c100369e0(Recorder/ARI-00000003;2) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is leaving softmix technology (dummy) [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is leaving softmix technology (dummy) [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology stop [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13315] res_ari.c: No explicit handler found for 378d72c1-dd9d-472b-9f36-5c575a6102e6. Using wildcard bridgeId. [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Finding handler for play [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel Recorder/ARI-00000003;2 setting read format path: slin -> slin [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:52] DEBUG[13315] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13315] stasis.c: Creating topic. name: channel:1629282832.89, detail: [Aug 18 10:33:52] DEBUG[13315] stasis.c: Topic 'channel:1629282832.89': 0x7f0c2808cdb0 created [Aug 18 10:33:52] DEBUG[13224] stasis.c: Destroying topic. name: cache:74/channel:1629282830.63, detail: [Aug 18 10:33:52] DEBUG[13224] stasis.c: Topic 'cache:74/channel:1629282830.63': 0x7f0c2c077310 destroyed [Aug 18 10:33:52] DEBUG[13224] stasis.c: Destroying topic. name: channel:1629282830.63, detail: [Aug 18 10:33:52] DEBUG[13224] stasis.c: Topic 'channel:1629282830.63': 0x7f0c2c01b1b0 destroyed [Aug 18 10:33:52] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:52] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:52] DEBUG[13315] stasis.c: Creating topic. name: cache:104/channel:1629282832.89, detail: [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel Recorder/ARI-00000003;2 setting read format path: slin -> slin [Aug 18 10:33:52] DEBUG[13315] stasis.c: Topic 'cache:104/channel:1629282832.89': 0x7f0c2808d730 created [Aug 18 10:33:52] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology start [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: deferring softmix technology destructor [Aug 18 10:33:52] DEBUG[13222] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: queueing action type:13 sub:1000 [Aug 18 10:33:52] DEBUG[13315] channel.c: Channel 0x7f0c280925d0 'Announcer/ARI-00000009;1' allocated [Aug 18 10:33:52] DEBUG[13315] stasis.c: Creating topic. name: channel:1629282832.90, detail: [Aug 18 10:33:52] DEBUG[13315] stasis.c: Topic 'channel:1629282832.90': 0x7f0c28010c40 created [Aug 18 10:33:52] DEBUG[13315] stasis.c: Creating topic. name: cache:105/channel:1629282832.90, detail: [Aug 18 10:33:52] DEBUG[13315] stasis.c: Topic 'cache:105/channel:1629282832.90': 0x7f0c28010e10 created [Aug 18 10:33:52] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:52] DEBUG[20534] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: Waiting for mixing thread to die. [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel 0x7f0c2c04ba30 'Announcer/ARI-00000004;2' hanging up. Refs: 2 [Aug 18 10:33:52] DEBUG[13222] channel.c: Channel 0x7f0c2c04ba30 'Announcer/ARI-00000004;2' destroying [Aug 18 10:33:52] DEBUG[13210] channel.c: Recorder/ARI-00000003;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:52] DEBUG[13208] channel.c: SIP/zvonobot-00000012: Dropping redundant connected line update "" <>. [Aug 18 10:33:52] DEBUG[13315] channel.c: Channel 0x7f0c28098390 'Announcer/ARI-00000009;2' allocated [Aug 18 10:33:52] DEBUG[13223] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: stopping mixing thread [Aug 18 10:33:52] DEBUG[13315] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:52] DEBUG[13222] stasis.c: Destroying topic. name: cache:76/channel:1629282830.65, detail: [Aug 18 10:33:52] DEBUG[13222] stasis.c: Topic 'cache:76/channel:1629282830.65': 0x7f0c2c00fcc0 destroyed [Aug 18 10:33:52] DEBUG[13222] stasis.c: Destroying topic. name: channel:1629282830.65, detail: [Aug 18 10:33:52] DEBUG[13315] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000009;1' [Aug 18 10:33:52] DEBUG[13222] stasis.c: Topic 'channel:1629282830.65': 0x7f0c2c00f460 destroyed [Aug 18 10:33:52] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:52] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:52] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:52] DEBUG[13316] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2800f490(Announcer/ARI-00000009;2) is joining [Aug 18 10:33:52] DEBUG[13316] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pushing 0x7f0c2800f490(Announcer/ARI-00000009;2) [Aug 18 10:33:52] DEBUG[13316] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:52] VERBOSE[13316] bridge_channel.c: Channel Announcer/ARI-00000009;2 joined 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13316] bridge.c: Chose bridge technology softmix [Aug 18 10:33:52] VERBOSE[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: switching from simple_bridge technology to softmix [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology constructor [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c080248a0(SIP/zvonobot-00000012) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c100369e0(Recorder/ARI-00000003;2) to dummy bridge temporarily [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is leaving simple_bridge technology (dummy) [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology stop [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2800f490(Announcer/ARI-00000009;2) is joining softmix technology [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Announcer/ARI-00000009;2: [Aug 18 10:33:52] DEBUG[13316] channel.c: Channel Announcer/ARI-00000009;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Announcer/ARI-00000009;2: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Announcer/ARI-00000009;2: Not in SFU mode [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is joining softmix technology [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: SIP/zvonobot-00000012: [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: SIP/zvonobot-00000012: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: SIP/zvonobot-00000012: Not in SFU mode [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining softmix technology [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Recorder/ARI-00000003;2: [Aug 18 10:33:52] DEBUG[13316] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:52] DEBUG[13316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Recorder/ARI-00000003;2: [Aug 18 10:33:52] DEBUG[13316] bridge_softmix.c: Recorder/ARI-00000003;2: Not in SFU mode [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology start [Aug 18 10:33:52] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology destructor [Aug 18 10:33:52] DEBUG[13317] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: starting mixing thread [Aug 18 10:33:52] DEBUG[13315] res_stasis_playback.c: 1629282832.89: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:52] DEBUG[13315] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:52] DEBUG[13315] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13318] channel.c: Channel Announcer/ARI-00000009;1 setting write format path: gsm -> slin [Aug 18 10:33:52] DEBUG[13318] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:52] VERBOSE[13318] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:52] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:52] DEBUG[13312] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 [Aug 18 10:33:52] DEBUG[13312] stasis/control.c: robot_212993: Adding to bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:33:52] DEBUG[13312] stasis/app.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' is 3 interested in calls_0 [Aug 18 10:33:52] DEBUG[13319] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) is joining [Aug 18 10:33:52] DEBUG[13319] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: pushing 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) [Aug 18 10:33:52] VERBOSE[13319] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 joined 'simple_bridge' stasis-bridge <9d1bf1e2-893f-4249-b006-4b3a345e76a2> [Aug 18 10:33:52] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 - start 1629282832.476723 answer 1629282832.479207 end 1629282832.685377 dur 0.208 bill 0.206 dispo ANSWERED [Aug 18 10:33:52] DEBUG[13319] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2'. Checking compatability for channels 'Snoop/212993-00000003' and 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20' [Aug 18 10:33:52] DEBUG[13319] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' can not use native RTP bridge as could not get details [Aug 18 10:33:52] DEBUG[13319] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:52] DEBUG[13319] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13319] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:52] DEBUG[13319] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:52] DEBUG[13319] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 is already using the new technology. [Aug 18 10:33:52] DEBUG[13319] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) is joining simple_bridge technology [Aug 18 10:33:52] DEBUG[13319] channel.c: Channel UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 setting read format path: slin16 -> slin16 [Aug 18 10:33:52] DEBUG[13319] channel.c: Channel Snoop/212993-00000003 setting write format path: slin16 -> slin [Aug 18 10:33:52] DEBUG[13319] channel.c: Channel Snoop/212993-00000003 setting read format path: slin -> slin16 [Aug 18 10:33:52] DEBUG[13319] channel.c: Channel UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 setting write format path: slin16 -> slin16 [Aug 18 10:33:52] DEBUG[13312] stasis/app.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' is 4 interested in calls_0 [Aug 18 10:33:52] DEBUG[13313] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:52] DEBUG[13313] http.c: HTTP closing session. Top level [Aug 18 10:33:52] DEBUG[13319] res_rtp_asterisk.c: (0x7f0c10045b20) RTP ooh, format changed from none to slin16 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b7e147e;received=159.65.48.104 From: ;tag=as4d3d785f To: ;tag=as0d3ccf68 Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 2092775894 2092775894 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12912 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b7e147e;received=159.65.48.104 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d3d785f [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0d3ccf68 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 2092775894 2092775894 IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12912 RTP/AVP 0 8 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 (Checking To) --From tag as4d3d785f --To-tag as0d3ccf68 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Stopping retransmission on '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Got SDP version 2092775894 and unique parts [root 2092775894 IN IP4 178.62.121.41] [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 2092775894 2092775894 IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE set role failed; no ice instance [Aug 18 10:33:53] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP setting address on RTP instance [Aug 18 10:33:53] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0cb003cb10 -- Strict RTP learning after remote address set to: 178.62.121.41:12912 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12912 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0091af8) from 0x7f0c147e2330 to 0x7f0cb0015d98 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00908d8) from 0x7f0c147e2330 to 0x7f0cb0015d98 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00779b8) from 0x7f0c147e2330 to 0x7f0cb0015d98 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP ignoring duplicate property [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000010 setting read format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000010 setting write format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS - ast_rtp_activate rtp=0x7f0cb003cb10 - setup and perform DTLS' [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb003cb10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb003cb10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:53] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Strict routing enforced for session 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117059@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcbcee7 Max-Forwards: 70 From: ;tag=as4d3d785f To: ;tag=as0d3ccf68 Contact: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:53] VERBOSE[12962] dial.c: SIP/zvonobot-00000010 answered [Aug 18 10:33:53] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:53] VERBOSE[12962] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000010 [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session timer started: 5 - 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 1768000ms [Aug 18 10:33:53] DEBUG[12962] stasis/app.c: Channel '212981' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:53] DEBUG[13320] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13320] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13320] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13320] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13320] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13320] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13320] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13320] stasis.c: Creating topic. name: bridge:0ec77f0c-7a86-4072-a1c4-e42f5256208c, detail: [Aug 18 10:33:53] DEBUG[13320] stasis.c: Topic 'bridge:0ec77f0c-7a86-4072-a1c4-e42f5256208c': 0x7f0c4000a200 created [Aug 18 10:33:53] DEBUG[13320] stasis.c: Creating topic. name: cache:106/bridge:0ec77f0c-7a86-4072-a1c4-e42f5256208c, detail: [Aug 18 10:33:53] DEBUG[13320] stasis.c: Topic 'cache:106/bridge:0ec77f0c-7a86-4072-a1c4-e42f5256208c': 0x7f0c40023cf0 created [Aug 18 10:33:53] DEBUG[13320] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13320] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13320] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13320] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13320] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13320] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13320] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13320] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13320] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13321] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13321] http.c: HTTP Request URI is /ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/addChannel?channel=212981 [Aug 18 10:33:53] DEBUG[13321] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13321] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13321] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13321] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Finding handler for bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/addChannel [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Finding handler for 0ec77f0c-7a86-4072-a1c4-e42f5256208c [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13321] res_ari.c: No explicit handler found for 0ec77f0c-7a86-4072-a1c4-e42f5256208c. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13321] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13321] stasis/control.c: 212981: Sending channel add_to_bridge command [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15c006b9;received=159.65.48.104 From: ;tag=as181bb145 To: ;tag=as4e94a883 Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1545699378 1545699378 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16138 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15c006b9;received=159.65.48.104 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as181bb145 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4e94a883 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1545699378 1545699378 IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16138 RTP/AVP 0 8 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 (Checking To) --From tag as181bb145 --To-tag as4e94a883 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Stopping retransmission on '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Got SDP version 1545699378 and unique parts [root 1545699378 IN IP4 178.62.121.41] [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1545699378 1545699378 IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) ICE set role failed; no ice instance [Aug 18 10:33:53] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) RTCP setting address on RTP instance [Aug 18 10:33:53] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c4000d2b0 -- Strict RTP learning after remote address set to: 178.62.121.41:16138 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:16138 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0044768) from 0x7f0c147e2330 to 0x7f0c40006528 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0043418) from 0x7f0c147e2330 to 0x7f0c40006528 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0024cb8) from 0x7f0c147e2330 to 0x7f0c40006528 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) RTCP ignoring duplicate property [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000008 setting read format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000008 setting write format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS - ast_rtp_activate rtp=0x7f0c4000d2b0 - setup and perform DTLS' [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4000d2b0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4000d2b0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:53] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117068@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f9831a8 Max-Forwards: 70 From: ;tag=as181bb145 To: ;tag=as4e94a883 Contact: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session timer started: 7 - 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 1768000ms [Aug 18 10:33:53] VERBOSE[12900] dial.c: SIP/zvonobot-00000008 answered [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:53] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:53] VERBOSE[12900] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000008 [Aug 18 10:33:53] DEBUG[12900] stasis/app.c: Channel '212972' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:53] DEBUG[13322] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13322] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13322] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13322] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13322] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13322] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13322] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13322] stasis.c: Creating topic. name: bridge:d36cece3-ab54-488a-bcb0-0ed40691a344, detail: [Aug 18 10:33:53] DEBUG[13322] stasis.c: Topic 'bridge:d36cece3-ab54-488a-bcb0-0ed40691a344': 0x7f0c70030fe0 created [Aug 18 10:33:53] DEBUG[13322] stasis.c: Creating topic. name: cache:107/bridge:d36cece3-ab54-488a-bcb0-0ed40691a344, detail: [Aug 18 10:33:53] DEBUG[13322] stasis.c: Topic 'cache:107/bridge:d36cece3-ab54-488a-bcb0-0ed40691a344': 0x7f0c7004a970 created [Aug 18 10:33:53] DEBUG[13322] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13322] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13322] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13322] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13322] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13322] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13322] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13322] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13322] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13323] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13323] http.c: HTTP Request URI is /ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/addChannel?channel=212972 [Aug 18 10:33:53] DEBUG[13323] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13323] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13323] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13323] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Finding handler for bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/addChannel [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Finding handler for d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13323] res_ari.c: No explicit handler found for d36cece3-ab54-488a-bcb0-0ed40691a344. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13323] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13323] stasis/control.c: 212972: Sending channel add_to_bridge command [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fbb5225;received=159.65.48.104 From: ;tag=as0453a0d2 To: ;tag=as6be15af9 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 262 v=0 o=root 69641570 69641570 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 19990 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fbb5225;received=159.65.48.104 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0453a0d2 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6be15af9 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 1 [ 45]: o=root 69641570 69641570 IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 19990 RTP/AVP 0 8 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking To) --From tag as0453a0d2 --To-tag as6be15af9 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Stopping retransmission on '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Got SDP version 69641570 and unique parts [root 69641570 IN IP4 178.62.121.41] [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 69641570 69641570 IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) ICE set role failed; no ice instance [Aug 18 10:33:53] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) RTCP setting address on RTP instance [Aug 18 10:33:53] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c70015c10 -- Strict RTP learning after remote address set to: 178.62.121.41:19990 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:19990 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0086e38) from 0x7f0c147e2330 to 0x7f0c70012358 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00250e8) from 0x7f0c147e2330 to 0x7f0c70012358 [Aug 18 10:33:53] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0044428) from 0x7f0c147e2330 to 0x7f0c70012358 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) RTCP ignoring duplicate property [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000009 setting read format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000009 setting write format path: alaw -> alaw [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS - ast_rtp_activate rtp=0x7f0c70015c10 - setup and perform DTLS' [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70015c10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70015c10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:53] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Strict routing enforced for session 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:53] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:53] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:53] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117067@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ee06140 Max-Forwards: 70 From: ;tag=as0453a0d2 To: ;tag=as6be15af9 Contact: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:53] DEBUG[20585] chan_sip.c: Session timer started: 11 - 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 1768000ms [Aug 18 10:33:53] VERBOSE[12903] dial.c: SIP/zvonobot-00000009 answered [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:53] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:53] VERBOSE[12903] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000009 [Aug 18 10:33:53] DEBUG[12903] stasis/app.c: Channel '212973' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:53] DEBUG[13324] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13324] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13324] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13324] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13324] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13324] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13324] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13324] stasis.c: Creating topic. name: bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e, detail: [Aug 18 10:33:53] DEBUG[13324] stasis.c: Topic 'bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e': 0x7f0c7802d570 created [Aug 18 10:33:53] DEBUG[13324] stasis.c: Creating topic. name: cache:108/bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e, detail: [Aug 18 10:33:53] DEBUG[13324] stasis.c: Topic 'cache:108/bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e': 0x7f0c7801a060 created [Aug 18 10:33:53] DEBUG[13324] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13324] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13324] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13324] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13324] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13324] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13324] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13324] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13324] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13325] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13325] http.c: HTTP Request URI is /ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/addChannel?channel=212973 [Aug 18 10:33:53] DEBUG[13325] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13325] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13325] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13325] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Finding handler for bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/addChannel [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Finding handler for 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13325] res_ari.c: No explicit handler found for 85c47f7b-0e24-4408-b7bf-5a532802bd8e. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13325] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13325] stasis/control.c: 212973: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13326] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13326] http.c: HTTP Request URI is /ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:53] DEBUG[13326] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13326] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13326] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13326] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Finding handler for bridges/90245cdf-0ee9-4414-b99e-c22349f119a2/play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Finding handler for 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13326] res_ari.c: No explicit handler found for 90245cdf-0ee9-4414-b99e-c22349f119a2. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Finding handler for play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:53] DEBUG[13326] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:53] DEBUG[13326] res_stasis_playback.c: 1629282831.70: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:53] DEBUG[13326] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13326] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13247] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:53] DEBUG[13247] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:53] DEBUG[13247] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:53] DEBUG[13247] channel.c: Channel Announcer/ARI-00000006;1 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13247] channel.c: Channel Announcer/ARI-00000006;1 setting write format path: gsm -> slin [Aug 18 10:33:53] DEBUG[13247] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:53] VERBOSE[13247] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:53] DEBUG[12962] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000010 [Aug 18 10:33:53] DEBUG[12962] stasis/control.c: 212981: Adding to bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c [Aug 18 10:33:53] DEBUG[12962] stasis/app.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13327] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0cac01e580(SIP/zvonobot-00000010) is joining [Aug 18 10:33:53] DEBUG[13327] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pushing 0x7f0cac01e580(SIP/zvonobot-00000010) [Aug 18 10:33:53] VERBOSE[13327] bridge_channel.c: Channel SIP/zvonobot-00000010 joined 'simple_bridge' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:33:53] DEBUG[13327] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c is already using the new technology. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0cac01e580(SIP/zvonobot-00000010) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP changing ssrc from 1481064168 to 738508993 due to a source change [Aug 18 10:33:53] DEBUG[12962] stasis/app.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[13321] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:53] DEBUG[13321] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13328] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13328] http.c: HTTP Request URI is /ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/record?name=212981_aQrjGvPalDIqbwcnfKZNLcuucqdyMhtK&format=wav [Aug 18 10:33:53] DEBUG[13328] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/record] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13328] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/record] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13328] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/record] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13328] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Finding handler for bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Finding handler for 0ec77f0c-7a86-4072-a1c4-e42f5256208c [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13328] res_ari.c: No explicit handler found for 0ec77f0c-7a86-4072-a1c4-e42f5256208c. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Finding handler for record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:53] DEBUG[13328] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:53] DEBUG[13328] stasis.c: Creating topic. name: channel:1629282833.91, detail: [Aug 18 10:33:53] DEBUG[13328] stasis.c: Topic 'channel:1629282833.91': 0x7f0c88042730 created [Aug 18 10:33:53] DEBUG[13328] stasis.c: Creating topic. name: cache:109/channel:1629282833.91, detail: [Aug 18 10:33:53] DEBUG[13328] stasis.c: Topic 'cache:109/channel:1629282833.91': 0x7f0c880428f0 created [Aug 18 10:33:53] DEBUG[13328] channel.c: Channel 0x7f0c88037560 'Recorder/ARI-0000000a;1' allocated [Aug 18 10:33:53] DEBUG[13328] stasis.c: Creating topic. name: channel:1629282833.92, detail: [Aug 18 10:33:53] DEBUG[13328] stasis.c: Topic 'channel:1629282833.92': 0x7f0c88049ad0 created [Aug 18 10:33:53] DEBUG[13328] stasis.c: Creating topic. name: cache:110/channel:1629282833.92, detail: [Aug 18 10:33:53] DEBUG[13328] stasis.c: Topic 'cache:110/channel:1629282833.92': 0x7f0c88049ce0 created [Aug 18 10:33:53] DEBUG[13328] channel.c: Channel 0x7f0c88047e20 'Recorder/ARI-0000000a;2' allocated [Aug 18 10:33:53] DEBUG[13328] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] DEBUG[13329] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is joining [Aug 18 10:33:53] DEBUG[13329] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pushing 0x7f0c88048a20(Recorder/ARI-0000000a;2) [Aug 18 10:33:53] DEBUG[13329] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] VERBOSE[13329] bridge_channel.c: Channel Recorder/ARI-0000000a;2 joined 'simple_bridge' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:33:53] DEBUG[13329] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c'. Checking compatability for channels 'SIP/zvonobot-00000010' and 'Recorder/ARI-0000000a;2' [Aug 18 10:33:53] DEBUG[13329] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as could not get details [Aug 18 10:33:53] DEBUG[13329] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13329] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13329] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13329] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13329] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c is already using the new technology. [Aug 18 10:33:53] DEBUG[13329] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13329] channel.c: Channel Recorder/ARI-0000000a;2 setting read format path: slin -> slin [Aug 18 10:33:53] DEBUG[13329] channel.c: Channel SIP/zvonobot-00000010 setting write format path: slin -> alaw [Aug 18 10:33:53] DEBUG[13329] channel.c: Channel SIP/zvonobot-00000010 setting read format path: alaw -> slin [Aug 18 10:33:53] DEBUG[13329] channel.c: Channel Recorder/ARI-0000000a;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13328] res_stasis_recording.c: 1629282833.91: Sending record(212981_aQrjGvPalDIqbwcnfKZNLcuucqdyMhtK.wav) command [Aug 18 10:33:53] DEBUG[13328] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13330] app.c: play_and_record: , /var/spool/asterisk/recording/212981_aQrjGvPalDIqbwcnfKZNLcuucqdyMhtK, 'wav' [Aug 18 10:33:53] DEBUG[13330] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:53] VERBOSE[13330] app.c: x=0, open writing: /var/spool/asterisk/recording/212981_aQrjGvPalDIqbwcnfKZNLcuucqdyMhtK format: wav, 0x7f0c90033c00 [Aug 18 10:33:53] DEBUG[13328] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13331] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13331] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13331] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13331] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13331] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13331] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13331] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13331] stasis.c: Creating topic. name: bridge:25e1770d-58e8-4da7-94aa-19844c10fa1c, detail: [Aug 18 10:33:53] DEBUG[13331] stasis.c: Topic 'bridge:25e1770d-58e8-4da7-94aa-19844c10fa1c': 0x7f0ca801d150 created [Aug 18 10:33:53] DEBUG[13331] stasis.c: Creating topic. name: cache:111/bridge:25e1770d-58e8-4da7-94aa-19844c10fa1c, detail: [Aug 18 10:33:53] DEBUG[13331] stasis.c: Topic 'cache:111/bridge:25e1770d-58e8-4da7-94aa-19844c10fa1c': 0x7f0ca800f440 created [Aug 18 10:33:53] DEBUG[13331] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13331] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13331] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13331] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13331] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13331] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13331] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13331] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13331] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13332] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13332] http.c: HTTP Request URI is /ari/channels/212981/snoop?app=calls_0&spy=in [Aug 18 10:33:53] DEBUG[13332] http.c: match request [ari/channels/212981/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13332] http.c: match request [ari/channels/212981/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13332] http.c: match request [ari/channels/212981/snoop] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13332] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Finding handler for channels/212981/snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Finding handler for 212981 [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channels create: Didn't match 212981 [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channels externalMedia: Didn't match 212981 [Aug 18 10:33:53] DEBUG[13332] res_ari.c: No explicit handler found for 212981. Using wildcard channelId. [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Finding handler for snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:53] DEBUG[13332] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:53] DEBUG[13332] stasis.c: Creating topic. name: channel:1629282833.93, detail: [Aug 18 10:33:53] DEBUG[13332] stasis.c: Topic 'channel:1629282833.93': 0x7f0c9c0321c0 created [Aug 18 10:33:53] DEBUG[13332] stasis.c: Creating topic. name: cache:112/channel:1629282833.93, detail: [Aug 18 10:33:53] DEBUG[13332] stasis.c: Topic 'cache:112/channel:1629282833.93': 0x7f0c9c032310 created [Aug 18 10:33:53] DEBUG[13332] channel.c: Channel 0x7f0c9c0305a0 'Snoop/212981-00000004' allocated [Aug 18 10:33:53] DEBUG[13327] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c'. Checking compatability for channels 'SIP/zvonobot-00000010' and 'Recorder/ARI-0000000a;2' [Aug 18 10:33:53] DEBUG[13327] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as channel 'SIP/zvonobot-00000010' has features which prevent it [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13327] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c is already using the new technology. [Aug 18 10:33:53] DEBUG[13333] stasis/app.c: Channel '1629282833.93' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 457 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 457 [Aug 18 10:33:53] DEBUG[13332] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13332] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13336] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13336] http.c: HTTP Request URI is /ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play?media=sound%3Asilence%2F2 [Aug 18 10:33:53] DEBUG[13336] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13336] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13336] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13336] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Finding handler for bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Finding handler for 0ec77f0c-7a86-4072-a1c4-e42f5256208c [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13336] res_ari.c: No explicit handler found for 0ec77f0c-7a86-4072-a1c4-e42f5256208c. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Finding handler for play [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13338] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:53] DEBUG[13336] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:53] DEBUG[13338] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212981&app=calls_0&format=slin16&external_host=127.0.0.1%3A50432 [Aug 18 10:33:53] DEBUG[13338] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13336] stasis.c: Creating topic. name: channel:1629282833.94, detail: [Aug 18 10:33:53] DEBUG[13338] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13338] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13338] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13336] stasis.c: Topic 'channel:1629282833.94': 0x7f0ca0035320 created [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:53] DEBUG[13336] stasis.c: Creating topic. name: cache:113/channel:1629282833.94, detail: [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13338] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:53] DEBUG[13338] netsock2.c: Splitting '127.0.0.1:50432' into... [Aug 18 10:33:53] DEBUG[13336] stasis.c: Topic 'cache:113/channel:1629282833.94': 0x7f0ca0041060 created [Aug 18 10:33:53] DEBUG[13338] netsock2.c: ...host '127.0.0.1' and port '50432'. [Aug 18 10:33:53] DEBUG[13338] netsock2.c: Splitting '127.0.0.1:50432' into... [Aug 18 10:33:53] DEBUG[13338] netsock2.c: ...host '127.0.0.1' and port '50432'. [Aug 18 10:33:53] DEBUG[13338] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] DEBUG[13338] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca406c1b0' [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP allocated port 17210 [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE creating session 127.0.0.1:17210 (17210) [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE create [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE add system candidates [Aug 18 10:33:53] DEBUG[13338] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:53] DEBUG[13336] channel.c: Channel 0x7f0ca003f150 'Announcer/ARI-0000000b;1' allocated [Aug 18 10:33:53] DEBUG[13338] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:53] DEBUG[13336] stasis.c: Creating topic. name: channel:1629282833.95, detail: [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE add candidate: 159.65.48.104:17210, 2130706431 [Aug 18 10:33:53] DEBUG[13338] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:53] DEBUG[13338] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:53] DEBUG[13336] stasis.c: Topic 'channel:1629282833.95': 0x7f0ca0048d00 created [Aug 18 10:33:53] DEBUG[13338] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE add candidate: 10.131.0.10:17210, 2130706431 [Aug 18 10:33:53] DEBUG[13336] stasis.c: Creating topic. name: cache:114/channel:1629282833.95, detail: [Aug 18 10:33:53] DEBUG[13338] rtp_engine.c: RTP instance '0x7f0ca406c1b0' is setup and ready to go [Aug 18 10:33:53] DEBUG[13338] stasis.c: Creating topic. name: channel:robot_212981, detail: [Aug 18 10:33:53] DEBUG[13338] stasis.c: Topic 'channel:robot_212981': 0x7f0ca40766b0 created [Aug 18 10:33:53] DEBUG[13338] stasis.c: Creating topic. name: cache:115/channel:robot_212981, detail: [Aug 18 10:33:53] DEBUG[13336] stasis.c: Topic 'cache:114/channel:1629282833.95': 0x7f0ca0046cd0 created [Aug 18 10:33:53] DEBUG[13338] stasis.c: Topic 'cache:115/channel:robot_212981': 0x7f0ca4073500 created [Aug 18 10:33:53] DEBUG[13336] channel.c: Channel 0x7f0ca0046fc0 'Announcer/ARI-0000000b;2' allocated [Aug 18 10:33:53] DEBUG[13336] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] DEBUG[13336] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000b;1' [Aug 18 10:33:53] DEBUG[13338] channel.c: Channel 0x7f0ca40752f0 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0' allocated [Aug 18 10:33:53] DEBUG[13340] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) is joining [Aug 18 10:33:53] DEBUG[13338] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] VERBOSE[13338] res_rtp_asterisk.c: 0x7f0ca406e190 -- Strict RTP learning after remote address set to: 127.0.0.1:50432 [Aug 18 10:33:53] DEBUG[13340] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pushing 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) [Aug 18 10:33:53] DEBUG[13338] res_stasis.c: calls_0: Subscribing to robot_212981 [Aug 18 10:33:53] DEBUG[13338] stasis/app.c: Channel 'robot_212981' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13338] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13338] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13340] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] VERBOSE[13340] bridge_channel.c: Channel Announcer/ARI-0000000b;2 joined 'simple_bridge' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:53] VERBOSE[13341] dial.c: Called 127.0.0.1:50432 [Aug 18 10:33:53] DEBUG[13340] bridge.c: Chose bridge technology softmix [Aug 18 10:33:53] VERBOSE[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: switching from simple_bridge technology to softmix [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling softmix technology constructor [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: moving 0x7f0cac01e580(SIP/zvonobot-00000010) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: moving 0x7f0c88048a20(Recorder/ARI-0000000a;2) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50432 [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0cac01e580(SIP/zvonobot-00000010) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology stop [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Announcer/ARI-0000000b;2: [Aug 18 10:33:53] DEBUG[13340] channel.c: Channel Announcer/ARI-0000000b;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: [Aug 18 10:33:53] VERBOSE[13341] dial.c: UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 answered [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Announcer/ARI-0000000b;2: [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Announcer/ARI-0000000b;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0cac01e580(SIP/zvonobot-00000010) is joining softmix technology [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: SIP/zvonobot-00000010: [Aug 18 10:33:53] VERBOSE[13341] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: SIP/zvonobot-00000010: [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: SIP/zvonobot-00000010: Not in SFU mode [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Recorder/ARI-0000000a;2: [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:53] DEBUG[13341] stasis/app.c: Channel 'robot_212981' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:53] DEBUG[13340] channel.c: Channel Recorder/ARI-0000000a;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13340] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Recorder/ARI-0000000a;2: [Aug 18 10:33:53] DEBUG[13340] bridge_softmix.c: Recorder/ARI-0000000a;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling softmix technology start [Aug 18 10:33:53] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology destructor [Aug 18 10:33:53] DEBUG[13343] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13343] http.c: HTTP Request URI is /ari/bridges/25e1770d-58e8-4da7-94aa-19844c10fa1c/addChannel?channel=1629282833.93%2Crobot_212981 [Aug 18 10:33:53] DEBUG[13343] http.c: match request [ari/bridges/25e1770d-58e8-4da7-94aa-19844c10fa1c/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13343] http.c: match request [ari/bridges/25e1770d-58e8-4da7-94aa-19844c10fa1c/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13342] bridge_softmix.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: starting mixing thread [Aug 18 10:33:53] DEBUG[13343] http.c: match request [ari/bridges/25e1770d-58e8-4da7-94aa-19844c10fa1c/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13336] res_stasis_playback.c: 1629282833.94: Sending play(sound:silence/2) command [Aug 18 10:33:53] DEBUG[13343] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Finding handler for bridges/25e1770d-58e8-4da7-94aa-19844c10fa1c/addChannel [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Finding handler for 25e1770d-58e8-4da7-94aa-19844c10fa1c [Aug 18 10:33:53] DEBUG[13336] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13343] res_ari.c: No explicit handler found for 25e1770d-58e8-4da7-94aa-19844c10fa1c. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP ooh, format changed from none to alaw [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13343] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP starting transmission [Aug 18 10:33:53] DEBUG[12900] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000008 [Aug 18 10:33:53] DEBUG[12900] stasis/control.c: 212972: Adding to bridge d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:33:53] DEBUG[12900] stasis/app.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13336] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13347] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is joining [Aug 18 10:33:53] VERBOSE[13327] res_rtp_asterisk.c: 0x7f0cb003cb10 -- Strict RTP switching to RTP target address 178.62.121.41:12912 as source [Aug 18 10:33:53] DEBUG[13347] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pushing 0x7f0c740224f0(SIP/zvonobot-00000008) [Aug 18 10:33:53] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] VERBOSE[13347] bridge_channel.c: Channel SIP/zvonobot-00000008 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:53] DEBUG[13347] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13346] channel.c: Channel Announcer/ARI-0000000b;1 setting write format path: gsm -> slin [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13347] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344 is already using the new technology. [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13346] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:53] VERBOSE[13346] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50432 - state 2 (In use) [Aug 18 10:33:53] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50432, detail: [Aug 18 10:33:53] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50432': 0x7f0c8404e940 created [Aug 18 10:33:53] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP changing ssrc from 616294050 to 953822116 due to a source change [Aug 18 10:33:53] DEBUG[12900] stasis/app.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50432' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:53] DEBUG[13323] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:53] DEBUG[13333] bridge_roles.c: Roles did not exist on channel Snoop/212981-00000004 [Aug 18 10:33:53] DEBUG[13343] stasis/control.c: 1629282833.93: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13323] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13348] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13348] http.c: HTTP Request URI is /ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/record?name=212972_aDKyhcVImNUpqwlLDxkaEFbchMLlEXUD&format=wav [Aug 18 10:33:53] DEBUG[13348] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/record] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13348] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/record] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13348] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/record] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13348] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Finding handler for bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Finding handler for d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13348] res_ari.c: No explicit handler found for d36cece3-ab54-488a-bcb0-0ed40691a344. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Finding handler for record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:53] DEBUG[13348] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:53] DEBUG[12903] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000009 [Aug 18 10:33:53] DEBUG[12903] stasis/control.c: 212973: Adding to bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:33:53] DEBUG[13348] stasis.c: Creating topic. name: channel:1629282833.97, detail: [Aug 18 10:33:53] DEBUG[12903] stasis/app.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13349] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is joining [Aug 18 10:33:53] DEBUG[13348] stasis.c: Topic 'channel:1629282833.97': 0x7f0c1c01e760 created [Aug 18 10:33:53] DEBUG[13348] stasis.c: Creating topic. name: cache:116/channel:1629282833.97, detail: [Aug 18 10:33:53] DEBUG[13349] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pushing 0x7f0c7c01ea60(SIP/zvonobot-00000009) [Aug 18 10:33:53] DEBUG[13348] stasis.c: Topic 'cache:116/channel:1629282833.97': 0x7f0c1c017750 created [Aug 18 10:33:53] VERBOSE[13349] bridge_channel.c: Channel SIP/zvonobot-00000009 joined 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:33:53] DEBUG[13349] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13349] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e is already using the new technology. [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13348] channel.c: Channel 0x7f0c1c012020 'Recorder/ARI-0000000c;1' allocated [Aug 18 10:33:53] DEBUG[13349] res_rtp_asterisk.c: (0x7f0c70012180) RTP changing ssrc from 1671207052 to 1668909859 due to a source change [Aug 18 10:33:53] DEBUG[12903] stasis/app.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[13325] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:53] DEBUG[13325] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13350] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13350] http.c: HTTP Request URI is /ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/record?name=212973_JMVUCcjsDWybTwNZgmRlAqMcmISvqRTX&format=wav [Aug 18 10:33:53] DEBUG[13348] stasis.c: Creating topic. name: channel:1629282833.98, detail: [Aug 18 10:33:53] DEBUG[13350] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/record] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13350] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/record] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13348] stasis.c: Topic 'channel:1629282833.98': 0x7f0c1c01de30 created [Aug 18 10:33:53] DEBUG[13350] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/record] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13350] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13348] stasis.c: Creating topic. name: cache:117/channel:1629282833.98, detail: [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Finding handler for bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13348] stasis.c: Topic 'cache:117/channel:1629282833.98': 0x7f0c1c01e050 created [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Finding handler for 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13350] res_ari.c: No explicit handler found for 85c47f7b-0e24-4408-b7bf-5a532802bd8e. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Finding handler for record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:53] DEBUG[13350] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:53] DEBUG[13350] stasis.c: Creating topic. name: channel:1629282833.99, detail: [Aug 18 10:33:53] DEBUG[13333] stasis/control.c: 1629282833.93: Adding to bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c [Aug 18 10:33:53] DEBUG[13333] stasis/app.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13350] stasis.c: Topic 'channel:1629282833.99': 0x7f0c24006850 created [Aug 18 10:33:53] DEBUG[13350] stasis.c: Creating topic. name: cache:118/channel:1629282833.99, detail: [Aug 18 10:33:53] DEBUG[13350] stasis.c: Topic 'cache:118/channel:1629282833.99': 0x7f0c24048530 created [Aug 18 10:33:53] DEBUG[13348] channel.c: Channel 0x7f0c1c04bed0 'Recorder/ARI-0000000c;2' allocated [Aug 18 10:33:53] DEBUG[13351] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0c9804a330(Snoop/212981-00000004) is joining [Aug 18 10:33:53] DEBUG[13351] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: pushing 0x7f0c9804a330(Snoop/212981-00000004) [Aug 18 10:33:53] DEBUG[13348] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] VERBOSE[13351] bridge_channel.c: Channel Snoop/212981-00000004 joined 'simple_bridge' stasis-bridge <25e1770d-58e8-4da7-94aa-19844c10fa1c> [Aug 18 10:33:53] DEBUG[13350] channel.c: Channel 0x7f0c240501f0 'Recorder/ARI-0000000d;1' allocated [Aug 18 10:33:53] DEBUG[13350] stasis.c: Creating topic. name: channel:1629282833.100, detail: [Aug 18 10:33:53] DEBUG[13350] stasis.c: Topic 'channel:1629282833.100': 0x7f0c240517f0 created [Aug 18 10:33:53] DEBUG[13350] stasis.c: Creating topic. name: cache:119/channel:1629282833.100, detail: [Aug 18 10:33:53] DEBUG[13350] stasis.c: Topic 'cache:119/channel:1629282833.100': 0x7f0c240588f0 created [Aug 18 10:33:53] DEBUG[13351] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13351] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13351] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13351] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13351] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13351] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c is already using the new technology. [Aug 18 10:33:53] DEBUG[13351] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0c9804a330(Snoop/212981-00000004) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13350] channel.c: Channel 0x7f0c24056340 'Recorder/ARI-0000000d;2' allocated [Aug 18 10:33:53] DEBUG[13352] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining [Aug 18 10:33:53] DEBUG[13343] stasis/control.c: robot_212981: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13333] stasis/app.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[13350] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] DEBUG[13352] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pushing 0x7f0c1c00f210(Recorder/ARI-0000000c;2) [Aug 18 10:33:53] DEBUG[13353] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining [Aug 18 10:33:53] DEBUG[13353] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pushing 0x7f0c240520b0(Recorder/ARI-0000000d;2) [Aug 18 10:33:53] DEBUG[13352] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] VERBOSE[13352] bridge_channel.c: Channel Recorder/ARI-0000000c;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:53] DEBUG[13353] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:53] VERBOSE[13353] bridge_channel.c: Channel Recorder/ARI-0000000d;2 joined 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:33:53] DEBUG[13352] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344'. Checking compatability for channels 'SIP/zvonobot-00000008' and 'Recorder/ARI-0000000c;2' [Aug 18 10:33:53] DEBUG[13352] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as could not get details [Aug 18 10:33:53] DEBUG[13352] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13352] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13352] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13352] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13352] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344 is already using the new technology. [Aug 18 10:33:53] DEBUG[13352] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13353] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e'. Checking compatability for channels 'SIP/zvonobot-00000009' and 'Recorder/ARI-0000000d;2' [Aug 18 10:33:53] DEBUG[13353] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as could not get details [Aug 18 10:33:53] DEBUG[13353] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13353] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13353] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13353] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13353] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e is already using the new technology. [Aug 18 10:33:53] DEBUG[13353] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13353] channel.c: Channel Recorder/ARI-0000000d;2 setting read format path: slin -> slin [Aug 18 10:33:53] DEBUG[13353] channel.c: Channel SIP/zvonobot-00000009 setting write format path: slin -> alaw [Aug 18 10:33:53] DEBUG[13352] channel.c: Channel Recorder/ARI-0000000c;2 setting read format path: slin -> slin [Aug 18 10:33:53] DEBUG[13352] channel.c: Channel SIP/zvonobot-00000008 setting write format path: slin -> alaw [Aug 18 10:33:53] DEBUG[13353] channel.c: Channel SIP/zvonobot-00000009 setting read format path: alaw -> slin [Aug 18 10:33:53] DEBUG[13353] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13352] channel.c: Channel SIP/zvonobot-00000008 setting read format path: alaw -> slin [Aug 18 10:33:53] DEBUG[13352] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13348] res_stasis_recording.c: 1629282833.97: Sending record(212972_aDKyhcVImNUpqwlLDxkaEFbchMLlEXUD.wav) command [Aug 18 10:33:53] DEBUG[13348] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13354] app.c: play_and_record: , /var/spool/asterisk/recording/212972_aDKyhcVImNUpqwlLDxkaEFbchMLlEXUD, 'wav' [Aug 18 10:33:53] DEBUG[13354] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:53] DEBUG[13348] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13350] res_stasis_recording.c: 1629282833.99: Sending record(212973_JMVUCcjsDWybTwNZgmRlAqMcmISvqRTX.wav) command [Aug 18 10:33:53] DEBUG[13350] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] VERBOSE[13354] app.c: x=0, open writing: /var/spool/asterisk/recording/212972_aDKyhcVImNUpqwlLDxkaEFbchMLlEXUD format: wav, 0x7f0c34004470 [Aug 18 10:33:53] DEBUG[13350] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13357] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13357] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13357] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13357] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13357] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13357] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13355] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13357] stasis.c: Creating topic. name: bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66, detail: [Aug 18 10:33:53] DEBUG[13355] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:53] DEBUG[13355] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13357] stasis.c: Topic 'bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66': 0x7f0c3803cb70 created [Aug 18 10:33:53] DEBUG[13357] stasis.c: Creating topic. name: cache:120/bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66, detail: [Aug 18 10:33:53] DEBUG[13357] stasis.c: Topic 'cache:120/bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66': 0x7f0c3803d560 created [Aug 18 10:33:53] DEBUG[13355] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13355] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13355] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13356] app.c: play_and_record: , /var/spool/asterisk/recording/212973_JMVUCcjsDWybTwNZgmRlAqMcmISvqRTX, 'wav' [Aug 18 10:33:53] DEBUG[13356] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:53] DEBUG[13357] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] VERBOSE[13356] app.c: x=0, open writing: /var/spool/asterisk/recording/212973_JMVUCcjsDWybTwNZgmRlAqMcmISvqRTX format: wav, 0x7f0c3005d210 [Aug 18 10:33:53] DEBUG[13357] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13357] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13357] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13357] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13355] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13355] stasis.c: Creating topic. name: bridge:4918ac35-38b0-4486-b626-7cf67dacf45b, detail: [Aug 18 10:33:53] DEBUG[13355] stasis.c: Topic 'bridge:4918ac35-38b0-4486-b626-7cf67dacf45b': 0x7f0c3c0305a0 created [Aug 18 10:33:53] DEBUG[13357] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13357] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13358] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13358] http.c: HTTP Request URI is /ari/channels/212973/snoop?app=calls_0&spy=in [Aug 18 10:33:53] DEBUG[13355] stasis.c: Creating topic. name: cache:121/bridge:4918ac35-38b0-4486-b626-7cf67dacf45b, detail: [Aug 18 10:33:53] DEBUG[13358] http.c: match request [ari/channels/212973/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13358] http.c: match request [ari/channels/212973/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13358] http.c: match request [ari/channels/212973/snoop] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13358] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13355] stasis.c: Topic 'cache:121/bridge:4918ac35-38b0-4486-b626-7cf67dacf45b': 0x7f0c3c009390 created [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Finding handler for channels/212973/snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Finding handler for 212973 [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channels create: Didn't match 212973 [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channels externalMedia: Didn't match 212973 [Aug 18 10:33:53] DEBUG[13355] bridge_native_rtp.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13358] res_ari.c: No explicit handler found for 212973. Using wildcard channelId. [Aug 18 10:33:53] DEBUG[13355] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Finding handler for snoop [Aug 18 10:33:53] DEBUG[13355] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:53] DEBUG[13355] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:53] DEBUG[13355] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:53] DEBUG[13355] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: calling simple_bridge technology constructor [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:53] DEBUG[13355] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: calling simple_bridge technology start [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:53] DEBUG[13355] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:53] DEBUG[13355] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13359] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13359] http.c: HTTP Request URI is /ari/channels/212972/snoop?app=calls_0&spy=in [Aug 18 10:33:53] DEBUG[13359] http.c: match request [ari/channels/212972/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13359] http.c: match request [ari/channels/212972/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13359] http.c: match request [ari/channels/212972/snoop] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13358] stasis.c: Creating topic. name: channel:1629282833.101, detail: [Aug 18 10:33:53] DEBUG[13359] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13358] stasis.c: Topic 'channel:1629282833.101': 0x7f0c40006720 created [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Finding handler for channels/212972/snoop [Aug 18 10:33:53] DEBUG[13358] stasis.c: Creating topic. name: cache:122/channel:1629282833.101, detail: [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13358] stasis.c: Topic 'cache:122/channel:1629282833.101': 0x7f0c40029460 created [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Finding handler for 212972 [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channels create: Didn't match 212972 [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channels externalMedia: Didn't match 212972 [Aug 18 10:33:53] DEBUG[13359] res_ari.c: No explicit handler found for 212972. Using wildcard channelId. [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Finding handler for snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:53] DEBUG[13358] channel.c: Channel 0x7f0c40046650 'Snoop/212973-00000005' allocated [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e'. Checking compatability for channels 'SIP/zvonobot-00000009' and 'Recorder/ARI-0000000d;2' [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as channel 'SIP/zvonobot-00000009' has features which prevent it [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13349] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:53] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e is already using the new technology. [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:53] DEBUG[13359] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:53] DEBUG[13359] stasis.c: Creating topic. name: channel:1629282833.102, detail: [Aug 18 10:33:53] DEBUG[13359] stasis.c: Topic 'channel:1629282833.102': 0x7f0c740229f0 created [Aug 18 10:33:53] DEBUG[13359] stasis.c: Creating topic. name: cache:123/channel:1629282833.102, detail: [Aug 18 10:33:53] DEBUG[13359] stasis.c: Topic 'cache:123/channel:1629282833.102': 0x7f0c74022cc0 created [Aug 18 10:33:53] DEBUG[13359] channel.c: Channel 0x7f0c74028ad0 'Snoop/212972-00000006' allocated [Aug 18 10:33:53] DEBUG[13358] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13361] stasis/app.c: Channel '1629282833.102' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13358] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13347] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344'. Checking compatability for channels 'SIP/zvonobot-00000008' and 'Recorder/ARI-0000000c;2' [Aug 18 10:33:53] DEBUG[13347] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as channel 'SIP/zvonobot-00000008' has features which prevent it [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13359] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:53] DEBUG[13347] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344 is already using the new technology. [Aug 18 10:33:53] DEBUG[13359] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13360] stasis/app.c: Channel '1629282833.101' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:53] DEBUG[13367] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13367] http.c: HTTP Request URI is /ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play?media=sound%3Asilence%2F2 [Aug 18 10:33:53] DEBUG[13367] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13367] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13367] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13367] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Finding handler for bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Finding handler for 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13367] res_ari.c: No explicit handler found for 85c47f7b-0e24-4408-b7bf-5a532802bd8e. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Finding handler for play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:53] DEBUG[13367] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:53] DEBUG[13370] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13367] stasis.c: Creating topic. name: channel:1629282833.103, detail: [Aug 18 10:33:53] DEBUG[13367] stasis.c: Topic 'channel:1629282833.103': 0x7f0c7803d710 created [Aug 18 10:33:53] DEBUG[13367] stasis.c: Creating topic. name: cache:124/channel:1629282833.103, detail: [Aug 18 10:33:53] DEBUG[13367] stasis.c: Topic 'cache:124/channel:1629282833.103': 0x7f0c7803e120 created [Aug 18 10:33:53] DEBUG[13370] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212972&app=calls_0&format=slin16&external_host=127.0.0.1%3A50283 [Aug 18 10:33:53] DEBUG[13373] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13373] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212973&app=calls_0&format=slin16&external_host=127.0.0.1%3A50139 [Aug 18 10:33:53] DEBUG[13370] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13370] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13370] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13370] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13368] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13368] http.c: HTTP Request URI is /ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play?media=sound%3Asilence%2F2 [Aug 18 10:33:53] DEBUG[13368] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:53] DEBUG[13368] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13368] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13368] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13373] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Finding handler for bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play [Aug 18 10:33:53] DEBUG[13373] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13373] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13373] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Finding handler for d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13370] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:53] DEBUG[13368] res_ari.c: No explicit handler found for d36cece3-ab54-488a-bcb0-0ed40691a344. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13370] netsock2.c: Splitting '127.0.0.1:50283' into... [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Finding handler for play [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13370] netsock2.c: ...host '127.0.0.1' and port '50283'. [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:53] DEBUG[13368] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:53] DEBUG[13370] netsock2.c: Splitting '127.0.0.1:50283' into... [Aug 18 10:33:53] DEBUG[13367] channel.c: Channel 0x7f0c7803b230 'Announcer/ARI-0000000e;1' allocated [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:53] DEBUG[13368] stasis.c: Creating topic. name: channel:1629282833.104, detail: [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Finding handler for channels [Aug 18 10:33:53] DEBUG[13370] netsock2.c: ...host '127.0.0.1' and port '50283'. [Aug 18 10:33:53] DEBUG[13367] stasis.c: Creating topic. name: channel:1629282833.105, detail: [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:53] DEBUG[13368] stasis.c: Topic 'channel:1629282833.104': 0x7f0c8002ddb0 created [Aug 18 10:33:53] DEBUG[13367] stasis.c: Topic 'channel:1629282833.105': 0x7f0c780398b0 created [Aug 18 10:33:53] DEBUG[13367] stasis.c: Creating topic. name: cache:126/channel:1629282833.105, detail: [Aug 18 10:33:53] DEBUG[13367] stasis.c: Topic 'cache:126/channel:1629282833.105': 0x7f0c7800a070 created [Aug 18 10:33:53] DEBUG[13370] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:53] DEBUG[13370] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c840529d0' [Aug 18 10:33:53] DEBUG[13367] channel.c: Channel 0x7f0c78042670 'Announcer/ARI-0000000e;2' allocated [Aug 18 10:33:53] DEBUG[13368] stasis.c: Creating topic. name: cache:125/channel:1629282833.104, detail: [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) RTP allocated port 18430 [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:53] DEBUG[13368] stasis.c: Topic 'cache:125/channel:1629282833.104': 0x7f0c8002e010 created [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) ICE creating session 127.0.0.1:18430 (18430) [Aug 18 10:33:53] DEBUG[13367] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) ICE create [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) ICE add system candidates [Aug 18 10:33:53] DEBUG[13367] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000e;1' [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13373] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:53] DEBUG[13373] netsock2.c: Splitting '127.0.0.1:50139' into... [Aug 18 10:33:53] DEBUG[13373] netsock2.c: ...host '127.0.0.1' and port '50139'. [Aug 18 10:33:53] DEBUG[13373] netsock2.c: Splitting '127.0.0.1:50139' into... [Aug 18 10:33:53] DEBUG[13373] netsock2.c: ...host '127.0.0.1' and port '50139'. [Aug 18 10:33:53] DEBUG[13373] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] DEBUG[13373] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c042660' [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) RTP allocated port 16196 [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) ICE creating session 127.0.0.1:16196 (16196) [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) ICE create [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) ICE add system candidates [Aug 18 10:33:53] DEBUG[13373] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:53] DEBUG[13373] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) ICE add candidate: 159.65.48.104:16196, 2130706431 [Aug 18 10:33:53] DEBUG[13373] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:53] DEBUG[13373] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:53] DEBUG[13373] res_rtp_asterisk.c: (0x7f0c8c042660) ICE add candidate: 10.131.0.10:16196, 2130706431 [Aug 18 10:33:53] DEBUG[13373] rtp_engine.c: RTP instance '0x7f0c8c042660' is setup and ready to go [Aug 18 10:33:53] DEBUG[13374] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7803ac60(Announcer/ARI-0000000e;2) is joining [Aug 18 10:33:53] DEBUG[13370] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:53] DEBUG[13370] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:53] DEBUG[13373] stasis.c: Creating topic. name: channel:robot_212973, detail: [Aug 18 10:33:53] DEBUG[13373] stasis.c: Topic 'channel:robot_212973': 0x7f0c8c059340 created [Aug 18 10:33:53] DEBUG[13373] stasis.c: Creating topic. name: cache:127/channel:robot_212973, detail: [Aug 18 10:33:53] DEBUG[13373] stasis.c: Topic 'cache:127/channel:robot_212973': 0x7f0c8c059550 created [Aug 18 10:33:53] DEBUG[13368] channel.c: Channel 0x7f0c80037a00 'Announcer/ARI-0000000f;1' allocated [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) ICE add candidate: 159.65.48.104:18430, 2130706431 [Aug 18 10:33:53] DEBUG[13368] stasis.c: Creating topic. name: channel:1629282833.107, detail: [Aug 18 10:33:53] DEBUG[13370] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:53] DEBUG[13370] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:53] DEBUG[13368] stasis.c: Topic 'channel:1629282833.107': 0x7f0c80041960 created [Aug 18 10:33:53] DEBUG[13368] stasis.c: Creating topic. name: cache:128/channel:1629282833.107, detail: [Aug 18 10:33:53] DEBUG[13370] res_rtp_asterisk.c: (0x7f0c840529d0) ICE add candidate: 10.131.0.10:18430, 2130706431 [Aug 18 10:33:53] DEBUG[13370] rtp_engine.c: RTP instance '0x7f0c840529d0' is setup and ready to go [Aug 18 10:33:53] DEBUG[13368] stasis.c: Topic 'cache:128/channel:1629282833.107': 0x7f0c80039ae0 created [Aug 18 10:33:53] DEBUG[13370] stasis.c: Creating topic. name: channel:robot_212972, detail: [Aug 18 10:33:53] DEBUG[13374] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pushing 0x7f0c7803ac60(Announcer/ARI-0000000e;2) [Aug 18 10:33:53] DEBUG[13370] stasis.c: Topic 'channel:robot_212972': 0x7f0c8405fd40 created [Aug 18 10:33:53] DEBUG[13373] channel.c: Channel 0x7f0c8c057560 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660' allocated [Aug 18 10:33:53] DEBUG[13370] stasis.c: Creating topic. name: cache:129/channel:robot_212972, detail: [Aug 18 10:33:53] DEBUG[13370] stasis.c: Topic 'cache:129/channel:robot_212972': 0x7f0c84062c40 created [Aug 18 10:33:53] DEBUG[13373] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] VERBOSE[13373] res_rtp_asterisk.c: 0x7f0c8c0519a0 -- Strict RTP learning after remote address set to: 127.0.0.1:50139 [Aug 18 10:33:53] DEBUG[13373] res_stasis.c: calls_0: Subscribing to robot_212973 [Aug 18 10:33:53] DEBUG[13373] stasis/app.c: Channel 'robot_212973' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13373] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[13368] channel.c: Channel 0x7f0c8003faf0 'Announcer/ARI-0000000f;2' allocated [Aug 18 10:33:53] DEBUG[13373] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13374] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] DEBUG[13368] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] VERBOSE[13374] bridge_channel.c: Channel Announcer/ARI-0000000e;2 joined 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:33:53] DEBUG[13368] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000f;1' [Aug 18 10:33:53] DEBUG[13376] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c80046b00(Announcer/ARI-0000000f;2) is joining [Aug 18 10:33:53] VERBOSE[13375] dial.c: Called 127.0.0.1:50139 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50139 [Aug 18 10:33:53] DEBUG[13370] channel.c: Channel 0x7f0c84060000 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0' allocated [Aug 18 10:33:53] DEBUG[13370] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:53] VERBOSE[13370] res_rtp_asterisk.c: 0x7f0c8405adf0 -- Strict RTP learning after remote address set to: 127.0.0.1:50283 [Aug 18 10:33:53] DEBUG[13376] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pushing 0x7f0c80046b00(Announcer/ARI-0000000f;2) [Aug 18 10:33:53] DEBUG[13370] res_stasis.c: calls_0: Subscribing to robot_212972 [Aug 18 10:33:53] DEBUG[13370] stasis/app.c: Channel 'robot_212972' is 1 interested in calls_0 [Aug 18 10:33:53] VERBOSE[13375] dial.c: UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 answered [Aug 18 10:33:53] DEBUG[13370] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50139 - state 2 (In use) [Aug 18 10:33:53] DEBUG[13370] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:53] DEBUG[13376] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:53] VERBOSE[13376] bridge_channel.c: Channel Announcer/ARI-0000000f;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:53] VERBOSE[13377] dial.c: Called 127.0.0.1:50283 [Aug 18 10:33:53] VERBOSE[13375] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 [Aug 18 10:33:53] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50139, detail: [Aug 18 10:33:53] VERBOSE[13377] dial.c: UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 answered [Aug 18 10:33:53] VERBOSE[13377] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 [Aug 18 10:33:53] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50139': 0x7f0c8403f920 created [Aug 18 10:33:53] DEBUG[13375] stasis/app.c: Channel 'robot_212973' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:53] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50139' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50283 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13376] bridge.c: Chose bridge technology softmix [Aug 18 10:33:53] VERBOSE[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: switching from simple_bridge technology to softmix [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology constructor [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c740224f0(SIP/zvonobot-00000008) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c1c00f210(Recorder/ARI-0000000c;2) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology stop [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c80046b00(Announcer/ARI-0000000f;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Announcer/ARI-0000000f;2: [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50283 - state 2 (In use) [Aug 18 10:33:53] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50283, detail: [Aug 18 10:33:53] DEBUG[13341] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 [Aug 18 10:33:53] DEBUG[13341] stasis/control.c: robot_212981: Adding to bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c [Aug 18 10:33:53] DEBUG[13376] channel.c: Channel Announcer/ARI-0000000f;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50283': 0x7f0c840685d0 created [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13341] stasis/app.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' is 3 interested in calls_0 [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Announcer/ARI-0000000f;2: [Aug 18 10:33:53] DEBUG[13377] stasis/app.c: Channel 'robot_212972' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Announcer/ARI-0000000f;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is joining softmix technology [Aug 18 10:33:53] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50283' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: SIP/zvonobot-00000008: [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13379] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) is joining [Aug 18 10:33:53] DEBUG[13379] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: pushing 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:53] VERBOSE[13379] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 joined 'simple_bridge' stasis-bridge <25e1770d-58e8-4da7-94aa-19844c10fa1c> [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13374] bridge.c: Chose bridge technology softmix [Aug 18 10:33:53] DEBUG[13379] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c'. Checking compatability for channels 'Snoop/212981-00000004' and 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0' [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13379] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' can not use native RTP bridge as could not get details [Aug 18 10:33:53] DEBUG[13379] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13379] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: SIP/zvonobot-00000008: [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: SIP/zvonobot-00000008: Not in SFU mode [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Recorder/ARI-0000000c;2: [Aug 18 10:33:53] DEBUG[13376] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13379] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13379] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13379] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c is already using the new technology. [Aug 18 10:33:53] DEBUG[13379] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13376] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Recorder/ARI-0000000c;2: [Aug 18 10:33:53] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 - start 1629282833.440381 answer 1629282833.453219 end 1629282833.668861 dur 0.228 bill 0.215 dispo ANSWERED [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 setting read format path: slin16 -> slin16 [Aug 18 10:33:53] DEBUG[13376] bridge_softmix.c: Recorder/ARI-0000000c;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology start [Aug 18 10:33:53] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology destructor [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel Snoop/212981-00000004 setting write format path: slin16 -> slin [Aug 18 10:33:53] VERBOSE[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: switching from simple_bridge technology to softmix [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology constructor [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel Snoop/212981-00000004 setting read format path: slin -> slin16 [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 setting write format path: slin16 -> slin16 [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c7c01ea60(SIP/zvonobot-00000009) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c240520b0(Recorder/ARI-0000000d;2) to dummy bridge temporarily [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology stop [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7803ac60(Announcer/ARI-0000000e;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Announcer/ARI-0000000e;2: [Aug 18 10:33:53] DEBUG[13374] channel.c: Channel Announcer/ARI-0000000e;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13343] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:53] DEBUG[13343] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13379] channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 setting write format path: slin -> slin16 [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Announcer/ARI-0000000e;2: [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Announcer/ARI-0000000e;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is joining softmix technology [Aug 18 10:33:53] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP ooh, format changed from none to slin16 [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: SIP/zvonobot-00000009: [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: SIP/zvonobot-00000009: [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: SIP/zvonobot-00000009: Not in SFU mode [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining softmix technology [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Recorder/ARI-0000000d;2: [Aug 18 10:33:53] DEBUG[13374] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:53] DEBUG[13374] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:53] DEBUG[13381] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13341] stasis/app.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' is 4 interested in calls_0 [Aug 18 10:33:53] DEBUG[13381] http.c: HTTP Request URI is /ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66/addChannel?channel=1629282833.101%2Crobot_212973 [Aug 18 10:33:53] DEBUG[13381] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13381] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13381] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13381] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Finding handler for bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66/addChannel [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Recorder/ARI-0000000d;2: [Aug 18 10:33:53] DEBUG[13374] bridge_softmix.c: Recorder/ARI-0000000d;2: Not in SFU mode [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology start [Aug 18 10:33:53] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology destructor [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13380] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: starting mixing thread [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13349] audiohook.c: Audiohook 0x7f0c40043a80 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] DEBUG[13349] res_rtp_asterisk.c: (0x7f0c70012180) RTP ooh, format changed from none to alaw [Aug 18 10:33:53] DEBUG[13349] res_rtp_asterisk.c: (0x7f0c70012180) RTCP starting transmission [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Finding handler for 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13381] res_ari.c: No explicit handler found for 6f6fd705-00c9-4b9f-a75f-d7c933b85e66. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13381] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13381] stasis/control.c: 1629282833.101: Sending channel add_to_bridge command [Aug 18 10:33:53] VERBOSE[13349] res_rtp_asterisk.c: 0x7f0c70015c10 -- Strict RTP switching to RTP target address 178.62.121.41:19990 as source [Aug 18 10:33:53] DEBUG[13349] audiohook.c: Audiohook 0x7f0c40043a80 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] DEBUG[13367] res_stasis_playback.c: 1629282833.103: Sending play(sound:silence/2) command [Aug 18 10:33:53] DEBUG[13360] bridge_roles.c: Roles did not exist on channel Snoop/212973-00000005 [Aug 18 10:33:53] DEBUG[13360] stasis/control.c: 1629282833.101: Adding to bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:33:53] DEBUG[13360] stasis/app.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13382] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c70055800(Snoop/212973-00000005) is joining [Aug 18 10:33:53] DEBUG[13382] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: pushing 0x7f0c70055800(Snoop/212973-00000005) [Aug 18 10:33:53] VERBOSE[13382] bridge_channel.c: Channel Snoop/212973-00000005 joined 'simple_bridge' stasis-bridge <6f6fd705-00c9-4b9f-a75f-d7c933b85e66> [Aug 18 10:33:53] DEBUG[13383] channel.c: Channel Announcer/ARI-0000000e;1 setting write format path: gsm -> slin [Aug 18 10:33:53] DEBUG[13382] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13382] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13382] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13382] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13382] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13382] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 is already using the new technology. [Aug 18 10:33:53] DEBUG[13382] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c70055800(Snoop/212973-00000005) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13367] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13383] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:53] VERBOSE[13383] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:53] DEBUG[13381] stasis/control.c: robot_212973: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13360] stasis/app.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' is 2 interested in calls_0 [Aug 18 10:33:53] DEBUG[13367] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:53] DEBUG[13368] res_stasis_playback.c: 1629282833.104: Sending play(sound:silence/2) command [Aug 18 10:33:53] DEBUG[13378] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: starting mixing thread [Aug 18 10:33:53] DEBUG[13368] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:53] DEBUG[13368] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] DEBUG[13384] channel.c: Channel Announcer/ARI-0000000f;1 setting write format path: gsm -> slin [Aug 18 10:33:53] DEBUG[13384] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:53] VERBOSE[13384] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:53] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP ooh, format changed from none to alaw [Aug 18 10:33:53] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTCP starting transmission [Aug 18 10:33:53] VERBOSE[13347] res_rtp_asterisk.c: 0x7f0c4000d2b0 -- Strict RTP switching to RTP target address 178.62.121.41:16138 as source [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:53] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:53] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:33:53] DEBUG[13375] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 [Aug 18 10:33:53] DEBUG[13375] stasis/control.c: robot_212973: Adding to bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:33:53] DEBUG[13375] stasis/app.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' is 3 interested in calls_0 [Aug 18 10:33:53] DEBUG[13385] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) is joining [Aug 18 10:33:53] DEBUG[13385] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: pushing 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) [Aug 18 10:33:53] VERBOSE[13385] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 joined 'simple_bridge' stasis-bridge <6f6fd705-00c9-4b9f-a75f-d7c933b85e66> [Aug 18 10:33:53] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 - start 1629282833.612002 answer 1629282833.633561 end 1629282833.857137 dur 0.245 bill 0.223 dispo ANSWERED [Aug 18 10:33:53] DEBUG[13385] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66'. Checking compatability for channels 'Snoop/212973-00000005' and 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660' [Aug 18 10:33:53] DEBUG[13385] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' can not use native RTP bridge as could not get details [Aug 18 10:33:53] DEBUG[13385] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13385] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13385] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13385] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13385] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 is already using the new technology. [Aug 18 10:33:53] DEBUG[13385] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13385] channel.c: Channel UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 setting read format path: slin16 -> slin16 [Aug 18 10:33:53] DEBUG[13385] channel.c: Channel Snoop/212973-00000005 setting write format path: slin16 -> slin [Aug 18 10:33:53] DEBUG[13385] channel.c: Channel Snoop/212973-00000005 setting read format path: slin -> slin16 [Aug 18 10:33:53] DEBUG[13385] channel.c: Channel UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 setting write format path: slin16 -> slin16 [Aug 18 10:33:53] DEBUG[13375] stasis/app.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' is 4 interested in calls_0 [Aug 18 10:33:53] DEBUG[13381] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:53] DEBUG[13381] http.c: HTTP closing session. Top level [Aug 18 10:33:53] DEBUG[13386] http.c: HTTP opening session. Top level [Aug 18 10:33:53] DEBUG[13386] http.c: HTTP Request URI is /ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b/addChannel?channel=1629282833.102%2Crobot_212972 [Aug 18 10:33:53] DEBUG[13386] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:53] DEBUG[13386] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:53] DEBUG[13386] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b/addChannel] with handler [ari] len 3 [Aug 18 10:33:53] DEBUG[13386] http.c: Match made with [ari] [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Finding handler for bridges/4918ac35-38b0-4486-b626-7cf67dacf45b/addChannel [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Finding handler for bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Finding handler for 4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:53] DEBUG[13386] res_ari.c: No explicit handler found for 4918ac35-38b0-4486-b626-7cf67dacf45b. Using wildcard bridgeId. [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Finding handler for addChannel [Aug 18 10:33:53] DEBUG[13386] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:53] DEBUG[13386] stasis/control.c: 1629282833.102: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13385] res_rtp_asterisk.c: (0x7f0c8c042660) RTP ooh, format changed from none to slin16 [Aug 18 10:33:53] DEBUG[13361] bridge_roles.c: Roles did not exist on channel Snoop/212972-00000006 [Aug 18 10:33:53] DEBUG[13361] stasis/control.c: 1629282833.102: Adding to bridge 4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:33:53] DEBUG[13361] stasis/app.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' is 1 interested in calls_0 [Aug 18 10:33:53] DEBUG[13387] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: 0x7f0c7c02d4b0(Snoop/212972-00000006) is joining [Aug 18 10:33:53] DEBUG[13387] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: pushing 0x7f0c7c02d4b0(Snoop/212972-00000006) [Aug 18 10:33:53] VERBOSE[13387] bridge_channel.c: Channel Snoop/212972-00000006 joined 'simple_bridge' stasis-bridge <4918ac35-38b0-4486-b626-7cf67dacf45b> [Aug 18 10:33:53] DEBUG[13387] bridge_native_rtp.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' can not use native RTP bridge as two channels are required [Aug 18 10:33:53] DEBUG[13387] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:53] DEBUG[13387] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13387] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:53] DEBUG[13387] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:53] DEBUG[13387] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b is already using the new technology. [Aug 18 10:33:53] DEBUG[13387] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: 0x7f0c7c02d4b0(Snoop/212972-00000006) is joining simple_bridge technology [Aug 18 10:33:53] DEBUG[13386] stasis/control.c: robot_212972: Sending channel add_to_bridge command [Aug 18 10:33:53] DEBUG[13361] stasis/app.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' is 2 interested in calls_0 [Aug 18 10:33:54] DEBUG[13389] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13389] http.c: HTTP Request URI is /ari/channels/213027?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117013&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13389] http.c: match request [ari/channels/213027] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13389] http.c: match request [ari/channels/213027] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13389] http.c: match request [ari/channels/213027] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13389] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13389] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Finding handler for channels/213027 [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Finding handler for 213027 [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking channels create: Didn't match 213027 [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13389] res_ari.c: Checking channels externalMedia: Didn't match 213027 [Aug 18 10:33:54] DEBUG[13389] res_ari.c: No explicit handler found for 213027. Using wildcard channelId. [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36;received=159.65.48.104 From: ;tag=as16e0fe9d To: ;tag=as21c49b51 Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as16e0fe9d [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as21c49b51 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 (Checking To) --From tag as16e0fe9d --To-tag as21c49b51 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117046@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK430b4e36 Max-Forwards: 70 From: ;tag=as16e0fe9d To: ;tag=as21c49b51 Contact: Call-ID: 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:54] VERBOSE[13105] dial.c: SIP/zvonobot-0000001e is busy [Aug 18 10:33:54] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001e - start 1629282828.006298 answer 0.000000 end 1629282834.038838 dur 6.032 bill 1629282834.038 dispo BUSY [Aug 18 10:33:54] DEBUG[13105] channel.c: Channel 0x7f0c7c015960 'SIP/zvonobot-0000001e' hanging up. Refs: 2 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:54] DEBUG[13105] chan_sip.c: Hangup call SIP/zvonobot-0000001e, SIP callid 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:54] DEBUG[13105] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] DEBUG[13105] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] DEBUG[13105] channel.c: Channel 0x7f0c7c015960 'SIP/zvonobot-0000001e' destroying [Aug 18 10:33:54] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282834.109, detail: [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:33:54] DEBUG[20620] stasis/app.c: channel '212994': is 0 interested in calls_0 [Aug 18 10:33:54] DEBUG[20620] stasis/app.c: channel '212994' unsubscribed from calls_0 [Aug 18 10:33:54] DEBUG[20545] stasis.c: Topic 'channel:1629282834.109': 0x7f0c3003bad0 created [Aug 18 10:33:54] DEBUG[20545] stasis.c: Creating topic. name: cache:130/channel:1629282834.109, detail: [Aug 18 10:33:54] DEBUG[20545] stasis.c: Topic 'cache:130/channel:1629282834.109': 0x7f0c30063540 created [Aug 18 10:33:54] DEBUG[20545] stasis.c: Destroying topic. name: cache:130/channel:1629282834.109, detail: [Aug 18 10:33:54] DEBUG[20545] stasis.c: Topic 'cache:130/channel:1629282834.109': 0x7f0c30063540 destroyed [Aug 18 10:33:54] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282834.109, detail: [Aug 18 10:33:54] DEBUG[20545] stasis.c: Topic 'channel:1629282834.109': 0x7f0c3003bad0 destroyed [Aug 18 10:33:54] DEBUG[13391] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13391] http.c: HTTP Request URI is /ari/channels/213024?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117016&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13391] http.c: match request [ari/channels/213024] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13391] http.c: match request [ari/channels/213024] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13391] http.c: match request [ari/channels/213024] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13391] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13391] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:48', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'BUSY', 3, '', '212994', '')] [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Finding handler for channels/213024 [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Finding handler for 213024 [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking channels create: Didn't match 213024 [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13391] res_ari.c: Checking channels externalMedia: Didn't match 213024 [Aug 18 10:33:54] DEBUG[13391] res_ari.c: No explicit handler found for 213024. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13377] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 [Aug 18 10:33:54] DEBUG[13377] stasis/control.c: robot_212972: Adding to bridge 4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:33:54] DEBUG[13377] stasis/app.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' is 3 interested in calls_0 [Aug 18 10:33:54] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13394] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) is joining [Aug 18 10:33:54] DEBUG[13395] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13105] stasis.c: Destroying topic. name: cache:46/channel:212994, detail: [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: Allocating new SIP dialog for 5e175e916b3779fd63bce3fe1538365f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13389] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c059c40' [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) RTP allocated port 12654 [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE creating session 0.0.0.0:12654 (12654) [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE create [Aug 18 10:33:54] DEBUG[13399] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE add system candidates [Aug 18 10:33:54] DEBUG[13389] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13389] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE add candidate: 159.65.48.104:12654, 2130706431 [Aug 18 10:33:54] DEBUG[13389] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13389] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13394] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: pushing 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) [Aug 18 10:33:54] DEBUG[13399] http.c: HTTP Request URI is /ari/channels/213025?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117015&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13395] http.c: HTTP Request URI is /ari/channels/213029?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117011&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13105] stasis.c: Topic 'cache:46/channel:212994': 0x7f0c7c0176e0 destroyed [Aug 18 10:33:54] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13105] stasis.c: Destroying topic. name: channel:212994, detail: [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE add candidate: 10.131.0.10:12654, 2130706431 [Aug 18 10:33:54] DEBUG[13389] rtp_engine.c: RTP instance '0x7f0c1c059c40' is setup and ready to go [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE stopped [Aug 18 10:33:54] DEBUG[13389] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13389] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13389] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13389] res_rtp_asterisk.c: (0x7f0c1c059c40) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13389] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13389] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: SIP call-id changed from '5e175e916b3779fd63bce3fe1538365f@127.0.1.1:5060' to '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13389] stasis.c: Creating topic. name: channel:213027, detail: [Aug 18 10:33:54] DEBUG[13389] stasis.c: Topic 'channel:213027': 0x7f0c1c060670 created [Aug 18 10:33:54] DEBUG[13389] stasis.c: Creating topic. name: cache:131/channel:213027, detail: [Aug 18 10:33:54] DEBUG[13389] stasis.c: Topic 'cache:131/channel:213027': 0x7f0c1c0608d0 created [Aug 18 10:33:54] DEBUG[13399] http.c: match request [ari/channels/213025] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13105] stasis.c: Topic 'channel:212994': 0x7f0c7c07c040 destroyed [Aug 18 10:33:54] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:54] DEBUG[13399] http.c: match request [ari/channels/213025] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13399] http.c: match request [ari/channels/213025] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:54] DEBUG[13399] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:54] DEBUG[13399] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Finding handler for channels/213025 [Aug 18 10:33:54] DEBUG[13395] http.c: match request [ari/channels/213029] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13403] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Finding handler for 213025 [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking channels create: Didn't match 213025 [Aug 18 10:33:54] VERBOSE[13394] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 joined 'simple_bridge' stasis-bridge <4918ac35-38b0-4486-b626-7cf67dacf45b> [Aug 18 10:33:54] DEBUG[13395] http.c: match request [ari/channels/213029] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13403] http.c: HTTP Request URI is /ari/channels/213026?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117014&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13399] res_ari.c: Checking channels externalMedia: Didn't match 213025 [Aug 18 10:33:54] DEBUG[13399] res_ari.c: No explicit handler found for 213025. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13403] http.c: match request [ari/channels/213026] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 - start 1629282833.634448 answer 1629282833.647195 end 1629282834.166693 dur 0.532 bill 0.519 dispo ANSWERED [Aug 18 10:33:54] DEBUG[13403] http.c: match request [ari/channels/213026] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13403] http.c: match request [ari/channels/213026] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13404] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13395] http.c: match request [ari/channels/213029] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13403] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13404] http.c: HTTP Request URI is /ari/channels/213030?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117010&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13403] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13384] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13395] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13395] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13394] bridge_native_rtp.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b'. Checking compatability for channels 'Snoop/212972-00000006' and 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0' [Aug 18 10:33:54] DEBUG[13394] bridge_native_rtp.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' can not use native RTP bridge as could not get details [Aug 18 10:33:54] DEBUG[13394] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:54] DEBUG[13394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[13394] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[13394] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:54] DEBUG[13394] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b is already using the new technology. [Aug 18 10:33:54] DEBUG[13394] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) is joining simple_bridge technology [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 setting read format path: slin16 -> slin16 [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel Snoop/212972-00000006 setting write format path: slin16 -> slin [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel Snoop/212972-00000006 setting read format path: slin -> slin16 [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 setting write format path: slin16 -> slin16 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Finding handler for channels/213026 [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Finding handler for channels/213029 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13404] http.c: match request [ari/channels/213030] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13404] http.c: match request [ari/channels/213030] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13405] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13405] http.c: HTTP Request URI is /ari/channels/213032?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117008&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13404] http.c: match request [ari/channels/213030] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Finding handler for 213026 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking channels create: Didn't match 213026 [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: Allocating new SIP dialog for 37f8a3c951f6594257b2198404f0cda6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13391] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c18094150' [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) RTP allocated port 11188 [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE creating session 0.0.0.0:11188 (11188) [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE create [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE add system candidates [Aug 18 10:33:54] DEBUG[13391] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13404] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13391] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE add candidate: 159.65.48.104:11188, 2130706431 [Aug 18 10:33:54] DEBUG[13391] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13391] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE add candidate: 10.131.0.10:11188, 2130706431 [Aug 18 10:33:54] DEBUG[13391] rtp_engine.c: RTP instance '0x7f0c18094150' is setup and ready to go [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) ICE stopped [Aug 18 10:33:54] DEBUG[13405] http.c: match request [ari/channels/213032] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13391] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13391] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13391] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13391] res_rtp_asterisk.c: (0x7f0c18094150) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13391] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13406] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13405] http.c: match request [ari/channels/213032] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13405] http.c: match request [ari/channels/213032] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13391] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13405] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13403] res_ari.c: Checking channels externalMedia: Didn't match 213026 [Aug 18 10:33:54] DEBUG[13403] res_ari.c: No explicit handler found for 213026. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13405] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: SIP call-id changed from '37f8a3c951f6594257b2198404f0cda6@127.0.1.1:5060' to '024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13391] stasis.c: Creating topic. name: channel:213024, detail: [Aug 18 10:33:54] DEBUG[13391] stasis.c: Topic 'channel:213024': 0x7f0c180a8fa0 created [Aug 18 10:33:54] DEBUG[13391] stasis.c: Creating topic. name: cache:132/channel:213024, detail: [Aug 18 10:33:54] DEBUG[13391] stasis.c: Topic 'cache:132/channel:213024': 0x7f0c180a9a20 created [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Finding handler for channels/213032 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13384] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13406] http.c: HTTP Request URI is /ari/channels/213028?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117012&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13406] http.c: match request [ari/channels/213028] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13406] http.c: match request [ari/channels/213028] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13406] http.c: match request [ari/channels/213028] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13406] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13377] stasis/app.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' is 4 interested in calls_0 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13386] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13386] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13406] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13408] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] http.c: HTTP Request URI is /ari/channels/212994 [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: Allocating new SIP dialog for 0cc0ef380ffbf22b1f17121f25fd8376@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Finding handler for 213032 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking channels create: Didn't match 213032 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13405] res_ari.c: Checking channels externalMedia: Didn't match 213032 [Aug 18 10:33:54] DEBUG[13405] res_ari.c: No explicit handler found for 213032. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:33:54] DEBUG[13408] http.c: HTTP Request URI is /ari/channels/213033?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117007&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13409] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[13409] http.c: HTTP Request URI is /ari/channels/213031?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117009&callerId=74950493843 [Aug 18 10:33:54] DEBUG[13409] http.c: match request [ari/channels/213031] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13409] http.c: match request [ari/channels/213031] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13409] http.c: match request [ari/channels/213031] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13409] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13409] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Finding handler for channels/213031 [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Finding handler for 213031 [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking channels create: Didn't match 213031 [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13409] res_ari.c: Checking channels externalMedia: Didn't match 213031 [Aug 18 10:33:54] DEBUG[13409] res_ari.c: No explicit handler found for 213031. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13399] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c008d30' [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) RTP allocated port 15984 [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE creating session 0.0.0.0:15984 (15984) [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE create [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE add system candidates [Aug 18 10:33:54] DEBUG[13399] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13399] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE add candidate: 159.65.48.104:15984, 2130706431 [Aug 18 10:33:54] DEBUG[13399] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13399] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE add candidate: 10.131.0.10:15984, 2130706431 [Aug 18 10:33:54] DEBUG[13399] rtp_engine.c: RTP instance '0x7f0c2c008d30' is setup and ready to go [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE stopped [Aug 18 10:33:54] DEBUG[13399] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13399] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13399] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13399] res_rtp_asterisk.c: (0x7f0c2c008d30) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13399] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13399] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: SIP call-id changed from '0cc0ef380ffbf22b1f17121f25fd8376@127.0.1.1:5060' to '0a35965008fea95b4665220a212af999@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13399] stasis.c: Creating topic. name: channel:213025, detail: [Aug 18 10:33:54] DEBUG[13399] stasis.c: Topic 'channel:213025': 0x7f0c2c069d40 created [Aug 18 10:33:54] DEBUG[13399] stasis.c: Creating topic. name: cache:133/channel:213025, detail: [Aug 18 10:33:54] DEBUG[13399] stasis.c: Topic 'cache:133/channel:213025': 0x7f0c2c06a7c0 created [Aug 18 10:33:54] DEBUG[13404] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Finding handler for channels/213030 [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:54] DEBUG[13408] http.c: match request [ari/channels/213033] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13408] http.c: match request [ari/channels/213033] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13408] http.c: match request [ari/channels/213033] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13408] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13408] http.c: HTTP consuming request body [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Finding handler for channels/213033 [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Finding handler for 213033 [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking channels create: Didn't match 213033 [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13408] res_ari.c: Checking channels externalMedia: Didn't match 213033 [Aug 18 10:33:54] DEBUG[13408] res_ari.c: No explicit handler found for 213033. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Finding handler for 213029 [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking channels create: Didn't match 213029 [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13395] res_ari.c: Checking channels externalMedia: Didn't match 213029 [Aug 18 10:33:54] DEBUG[13395] res_ari.c: No explicit handler found for 213029. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13407] http.c: match request [ari/channels/212994] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[13407] http.c: match request [ari/channels/212994] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13407] http.c: match request [ari/channels/212994] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13407] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Finding handler for channels/212994 [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Finding handler for 213030 [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking channels create: Didn't match 213030 [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13404] res_ari.c: Checking channels externalMedia: Didn't match 213030 [Aug 18 10:33:54] DEBUG[13404] res_ari.c: No explicit handler found for 213030. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13389] channel.c: Channel 0x7f0c1c05ef50 'SIP/zvonobot-0000003c' allocated [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Finding handler for 212994 [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking channels create: Didn't match 212994 [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13407] res_ari.c: Checking channels externalMedia: Didn't match 212994 [Aug 18 10:33:54] DEBUG[13407] res_ari.c: No explicit handler found for 212994. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13389] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Finding handler for channels/213028 [Aug 18 10:33:54] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Destroying SIP dialog 33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '33db5d3e7ee912df36ba27e0630ae543@159.65.48.104:5060' Method: INVITE [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS stop [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:54] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c00e8e0) ICE RTP transport deallocating [Aug 18 10:33:54] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7c00e8e0' [Aug 18 10:33:54] DEBUG[13389] res_stasis.c: calls_0: Subscribing to 213027 [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Finding handler for channels [Aug 18 10:33:54] DEBUG[13389] stasis/app.c: Channel '213027' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:54] DEBUG[13394] channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 setting write format path: slin -> slin16 [Aug 18 10:33:54] DEBUG[13389] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Finding handler for 213028 [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking channels create: Didn't match 213028 [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:54] DEBUG[13406] res_ari.c: Checking channels externalMedia: Didn't match 213028 [Aug 18 10:33:54] DEBUG[13406] res_ari.c: No explicit handler found for 213028. Using wildcard channelId. [Aug 18 10:33:54] DEBUG[13389] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13311] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:54] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP ooh, format changed from none to slin16 [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: Allocating new SIP dialog for 65b7b7642630b661427d6cdc219248e3@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13405] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c30074490' [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) RTP allocated port 15126 [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE creating session 0.0.0.0:15126 (15126) [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE create [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE add system candidates [Aug 18 10:33:54] DEBUG[13405] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13405] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE add candidate: 159.65.48.104:15126, 2130706431 [Aug 18 10:33:54] DEBUG[13405] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13405] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE add candidate: 10.131.0.10:15126, 2130706431 [Aug 18 10:33:54] DEBUG[13405] rtp_engine.c: RTP instance '0x7f0c30074490' is setup and ready to go [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) ICE stopped [Aug 18 10:33:54] DEBUG[13405] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13405] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13405] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13405] res_rtp_asterisk.c: (0x7f0c30074490) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13405] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Outgoing Call for 79821117013 [Aug 18 10:33:54] DEBUG[13405] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: SIP call-id changed from '65b7b7642630b661427d6cdc219248e3@127.0.1.1:5060' to '08a58ce821a53822322944fb39331df0@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13405] stasis.c: Creating topic. name: channel:213032, detail: [Aug 18 10:33:54] DEBUG[13405] stasis.c: Topic 'channel:213032': 0x7f0c300935a0 created [Aug 18 10:33:54] DEBUG[13405] stasis.c: Creating topic. name: cache:134/channel:213032, detail: [Aug 18 10:33:54] DEBUG[13405] stasis.c: Topic 'cache:134/channel:213032': 0x7f0c30094020 created [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13391] channel.c: Channel 0x7f0c180a7220 'SIP/zvonobot-0000003d' allocated [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13391] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13391] res_stasis.c: calls_0: Subscribing to 213024 [Aug 18 10:33:54] DEBUG[13391] stasis/app.c: Channel '213024' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13384] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Outgoing Call for 79821117016 [Aug 18 10:33:54] DEBUG[13391] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13391] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: Allocating new SIP dialog for 767af71e6283f895552440fc18479eb7@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13395] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2003ba40' [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) RTP allocated port 10922 [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE creating session 0.0.0.0:10922 (10922) [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE create [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE add system candidates [Aug 18 10:33:54] DEBUG[13395] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13395] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE add candidate: 159.65.48.104:10922, 2130706431 [Aug 18 10:33:54] DEBUG[13395] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13395] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE add candidate: 10.131.0.10:10922, 2130706431 [Aug 18 10:33:54] DEBUG[13395] rtp_engine.c: RTP instance '0x7f0c2003ba40' is setup and ready to go [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE stopped [Aug 18 10:33:54] DEBUG[13395] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13395] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13395] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13395] res_rtp_asterisk.c: (0x7f0c2003ba40) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13395] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13395] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: SIP call-id changed from '767af71e6283f895552440fc18479eb7@127.0.1.1:5060' to '543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13395] stasis.c: Creating topic. name: channel:213029, detail: [Aug 18 10:33:54] DEBUG[13395] stasis.c: Topic 'channel:213029': 0x7f0c2004d6a0 created [Aug 18 10:33:54] DEBUG[13395] stasis.c: Creating topic. name: cache:135/channel:213029, detail: [Aug 18 10:33:54] DEBUG[13395] stasis.c: Topic 'cache:135/channel:213029': 0x7f0c2004e120 created [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13411] chan_sip.c: Audio is at 11188 [Aug 18 10:33:54] VERBOSE[13410] chan_sip.c: Audio is at 12654 [Aug 18 10:33:54] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:54] VERBOSE[13410] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13411] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13410] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13311] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:33:54] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] VERBOSE[13411] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: Allocating new SIP dialog for 63c8cb704d8c71467d63c5d9154b87d6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13404] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3403eb10' [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) RTP allocated port 16012 [Aug 18 10:33:54] VERBOSE[13410] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] VERBOSE[13411] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE creating session 0.0.0.0:16012 (16012) [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Initializing initreq for method INVITE - callid 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117013@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK44fb7d4d [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 3 [ 52]: From: ;tag=as4e77dae5 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 6 [ 60]: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13410] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117013@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK44fb7d4d Max-Forwards: 70 From: ;tag=as4e77dae5 To: Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 269566349 269566349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12654 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19 [Aug 18 10:33:54] DEBUG[13410] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK44fb7d4d;received=159.65.48.104 From: ;tag=as4e77dae5 To: ;tag=as18dd1346 Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02a8a829" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK44fb7d4d;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4e77dae5 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as18dd1346 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02a8a829" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 (Checking To) --From tag as4e77dae5 --To-tag as18dd1346 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117013@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK44fb7d4d Max-Forwards: 70 From: ;tag=as4e77dae5 To: ;tag=as18dd1346 Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 12654 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117013@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222 Max-Forwards: 70 From: ;tag=as4e77dae5 To: Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117013@178.62.121.41", nonce="02a8a829", response="8597442f307b5cb6d6a8478a428df101" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 269566349 269566350 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12654 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE create [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE add system candidates [Aug 18 10:33:54] DEBUG[13404] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13404] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE add candidate: 159.65.48.104:16012, 2130706431 [Aug 18 10:33:54] DEBUG[13404] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13404] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE add candidate: 10.131.0.10:16012, 2130706431 [Aug 18 10:33:54] DEBUG[13404] rtp_engine.c: RTP instance '0x7f0c3403eb10' is setup and ready to go [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) ICE stopped [Aug 18 10:33:54] DEBUG[13404] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13404] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 From: ;tag=as4e77dae5 To: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[13404] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Initializing initreq for method INVITE - callid 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117016@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4a70ba08 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 3 [ 52]: From: ;tag=as1edcb3d8 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 6 [ 60]: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13404] res_rtp_asterisk.c: (0x7f0c3403eb10) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13404] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13404] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: SIP call-id changed from '63c8cb704d8c71467d63c5d9154b87d6@127.0.1.1:5060' to '4005320d167483a22fe0427650e34c89@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13404] stasis.c: Creating topic. name: channel:213030, detail: [Aug 18 10:33:54] DEBUG[13404] stasis.c: Topic 'channel:213030': 0x7f0c34098650 created [Aug 18 10:33:54] DEBUG[13404] stasis.c: Creating topic. name: cache:136/channel:213030, detail: [Aug 18 10:33:54] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: Allocating new SIP dialog for 1bd156516f9cfdaf7af16a920e507f23@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13403] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280b2c40' [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) RTP allocated port 18660 [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE creating session 0.0.0.0:18660 (18660) [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE create [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE add system candidates [Aug 18 10:33:54] VERBOSE[13410] dial.c: Called zvonobot/79821117013 [Aug 18 10:33:54] DEBUG[13404] stasis.c: Topic 'cache:136/channel:213030': 0x7f0c340990d0 created [Aug 18 10:33:54] DEBUG[13403] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13403] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13411] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117016@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4a70ba08 Max-Forwards: 70 From: ;tag=as1edcb3d8 To: Contact: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1142897947 1142897947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11188 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [Aug 18 10:33:54] DEBUG[13411] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE add candidate: 159.65.48.104:18660, 2130706431 [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: Allocating new SIP dialog for 2307937004d92a07142f91ec54a0ab0a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13408] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c40036750' [Aug 18 10:33:54] DEBUG[13412] http.c: HTTP opening session. Top level [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4e77dae5 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 (Checking To) --From tag as4e77dae5 --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #22 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13403] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) RTP allocated port 19310 [Aug 18 10:33:54] DEBUG[13403] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE creating session 0.0.0.0:19310 (19310) [Aug 18 10:33:54] DEBUG[13399] channel.c: Channel 0x7f0c2c067fc0 'SIP/zvonobot-0000003e' allocated [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4a70ba08;received=159.65.48.104 From: ;tag=as1edcb3d8 To: ;tag=as1b3438f5 Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48c2a642" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4a70ba08;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1edcb3d8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1b3438f5 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48c2a642" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 (Checking To) --From tag as1edcb3d8 --To-tag as1b3438f5 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117016@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4a70ba08 Max-Forwards: 70 From: ;tag=as1edcb3d8 To: ;tag=as1b3438f5 Contact: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 11188 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117016@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dff3038 Max-Forwards: 70 From: ;tag=as1edcb3d8 To: Contact: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117016@178.62.121.41", nonce="48c2a642", response="c26f21050718ad1b72668351b6b12c09" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1142897947 1142897948 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11188 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE create [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE add candidate: 10.131.0.10:18660, 2130706431 [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13399] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE add system candidates [Aug 18 10:33:54] DEBUG[13403] rtp_engine.c: RTP instance '0x7f0c280b2c40' is setup and ready to go [Aug 18 10:33:54] DEBUG[13408] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE stopped [Aug 18 10:33:54] DEBUG[13408] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE add candidate: 159.65.48.104:19310, 2130706431 [Aug 18 10:33:54] DEBUG[13408] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13408] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE add candidate: 10.131.0.10:19310, 2130706431 [Aug 18 10:33:54] DEBUG[13408] rtp_engine.c: RTP instance '0x7f0c40036750' is setup and ready to go [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) ICE stopped [Aug 18 10:33:54] DEBUG[13408] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13408] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13408] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13408] res_rtp_asterisk.c: (0x7f0c40036750) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13408] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13408] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13403] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13403] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13403] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13403] res_rtp_asterisk.c: (0x7f0c280b2c40) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13403] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13403] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] VERBOSE[13411] dial.c: Called zvonobot/79821117016 [Aug 18 10:33:54] DEBUG[13412] http.c: HTTP Request URI is /ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:54] DEBUG[13399] res_stasis.c: calls_0: Subscribing to 213025 [Aug 18 10:33:54] DEBUG[13399] stasis/app.c: Channel '213025' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13399] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13399] http.c: HTTP closing session. Top level [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dff3038;received=159.65.48.104 From: ;tag=as1edcb3d8 To: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dff3038;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Outgoing Call for 79821117015 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1edcb3d8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13311] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: SIP call-id changed from '2307937004d92a07142f91ec54a0ab0a@127.0.1.1:5060' to '10f580a044264908688c62534aa40882@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[13405] channel.c: Channel 0x7f0c30091820 'SIP/zvonobot-0000003f' allocated [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: SIP call-id changed from '1bd156516f9cfdaf7af16a920e507f23@127.0.1.1:5060' to '4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13408] stasis.c: Creating topic. name: channel:213033, detail: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 (Checking To) --From tag as1edcb3d8 --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #29 - INVITE (got response) [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13405] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[13412] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [httpstatus] len 10 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13403] stasis.c: Creating topic. name: channel:213026, detail: [Aug 18 10:33:54] DEBUG[13403] stasis.c: Topic 'channel:213026': 0x7f0c280bc980 created [Aug 18 10:33:54] DEBUG[13403] stasis.c: Creating topic. name: cache:137/channel:213026, detail: [Aug 18 10:33:54] DEBUG[13403] stasis.c: Topic 'cache:137/channel:213026': 0x7f0c280b8290 created [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13408] stasis.c: Topic 'channel:213033': 0x7f0c4004ea00 created [Aug 18 10:33:54] DEBUG[13408] stasis.c: Creating topic. name: cache:138/channel:213033, detail: [Aug 18 10:33:54] DEBUG[13408] stasis.c: Topic 'cache:138/channel:213033': 0x7f0c40059240 created [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13412] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [phoneprov] len 9 [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: Allocating new SIP dialog for 05d9bf22000f427c5edbcb4f7cae14fd@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13406] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c031e30' [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) RTP allocated port 16048 [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE creating session 0.0.0.0:16048 (16048) [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE create [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE add system candidates [Aug 18 10:33:54] DEBUG[13406] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[13406] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE add candidate: 159.65.48.104:16048, 2130706431 [Aug 18 10:33:54] DEBUG[13406] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13406] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE add candidate: 10.131.0.10:16048, 2130706431 [Aug 18 10:33:54] DEBUG[13406] rtp_engine.c: RTP instance '0x7f0c3c031e30' is setup and ready to go [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE stopped [Aug 18 10:33:54] DEBUG[13406] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13406] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[13406] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] DEBUG[13406] res_rtp_asterisk.c: (0x7f0c3c031e30) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[13406] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[13405] res_stasis.c: calls_0: Subscribing to 213032 [Aug 18 10:33:54] DEBUG[13405] stasis/app.c: Channel '213032' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13405] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Outgoing Call for 79821117008 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13406] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13416] chan_sip.c: Audio is at 15126 [Aug 18 10:33:54] DEBUG[13405] http.c: HTTP closing session. Top level [Aug 18 10:33:54] VERBOSE[13416] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] VERBOSE[13416] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13415] chan_sip.c: Audio is at 15984 [Aug 18 10:33:54] VERBOSE[13415] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] VERBOSE[13416] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] VERBOSE[13415] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] VERBOSE[13415] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13412] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play] with handler [ari] len 3 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Initializing initreq for method INVITE - callid 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117008@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: SIP call-id changed from '05d9bf22000f427c5edbcb4f7cae14fd@127.0.1.1:5060' to '72b609e44febd02a23174a4d311879ff@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[13406] stasis.c: Creating topic. name: channel:213028, detail: [Aug 18 10:33:54] DEBUG[13406] stasis.c: Topic 'channel:213028': 0x7f0c3c0956a0 created [Aug 18 10:33:54] DEBUG[13406] stasis.c: Creating topic. name: cache:139/channel:213028, detail: [Aug 18 10:33:54] DEBUG[13406] stasis.c: Topic 'cache:139/channel:213028': 0x7f0c3c0513c0 created [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: Allocating new SIP dialog for 7c5d855836c6e2024acb04592e67a9ce@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57a37ae1 [Aug 18 10:33:54] DEBUG[13412] http.c: Match made with [ari] [Aug 18 10:33:54] DEBUG[13409] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c74032f50' [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 3 [ 52]: From: ;tag=as11b813e8 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Finding handler for bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59/play [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Finding handler for bridges [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 6 [ 60]: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) RTP allocated port 14410 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE creating session 0.0.0.0:14410 (14410) [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Initializing initreq for method INVITE - callid 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117015@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE create [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74fe287d [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] VERBOSE[13416] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117008@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57a37ae1 Max-Forwards: 70 From: ;tag=as11b813e8 To: Contact: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1666076223 1666076223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15126 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 3 [ 52]: From: ;tag=as6658275e [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17 [Aug 18 10:33:54] DEBUG[13416] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE add system candidates [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57a37ae1;received=159.65.48.104 From: ;tag=as11b813e8 To: ;tag=as2226a7e6 Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6643ca38" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57a37ae1;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as11b813e8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2226a7e6 [Aug 18 10:33:54] VERBOSE[13416] dial.c: Called zvonobot/79821117008 [Aug 18 10:33:54] DEBUG[13409] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6643ca38" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 6 [ 60]: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 (Checking To) --From tag as11b813e8 --To-tag as2226a7e6 [Aug 18 10:33:54] DEBUG[13409] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #17 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '08a58ce821a53822322944fb39331df0@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117008@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57a37ae1 Max-Forwards: 70 From: ;tag=as11b813e8 To: ;tag=as2226a7e6 Contact: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE add candidate: 159.65.48.104:14410, 2130706431 [Aug 18 10:33:54] DEBUG[13409] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:54] DEBUG[13409] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE add candidate: 10.131.0.10:14410, 2130706431 [Aug 18 10:33:54] DEBUG[13409] rtp_engine.c: RTP instance '0x7f0c74032f50' is setup and ready to go [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) ICE stopped [Aug 18 10:33:54] DEBUG[13409] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13409] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Finding handler for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 15126 [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[13409] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13412] res_ari.c: No explicit handler found for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59. Using wildcard bridgeId. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117008@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481 Max-Forwards: 70 From: ;tag=as11b813e8 To: Contact: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117008@178.62.121.41", nonce="6643ca38", response="228fbefed627b7efd3d0334f90d68bd3" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1666076223 1666076224 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15126 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Finding handler for play [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:54] DEBUG[13409] res_rtp_asterisk.c: (0x7f0c74032f50) RTCP setup on RTP instance [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 From: ;tag=as11b813e8 To: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:54] VERBOSE[13409] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as11b813e8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] VERBOSE[13415] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117015@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74fe287d Max-Forwards: 70 From: ;tag=as6658275e To: Contact: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 260 v=0 o=root 1263250 1263250 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15984 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13412] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 (Checking To) --From tag as11b813e8 --To-tag [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '08a58ce821a53822322944fb39331df0@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13415] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74fe287d;received=159.65.48.104 From: ;tag=as6658275e To: ;tag=as486daec8 Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6efa756d" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74fe287d;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6658275e [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as486daec8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6efa756d" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 (Checking To) --From tag as6658275e --To-tag as486daec8 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '0a35965008fea95b4665220a212af999@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117015@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74fe287d Max-Forwards: 70 From: ;tag=as6658275e To: ;tag=as486daec8 Contact: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 15984 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] VERBOSE[13415] dial.c: Called zvonobot/79821117015 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117015@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK435e0a94 Max-Forwards: 70 From: ;tag=as6658275e To: Contact: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117015@178.62.121.41", nonce="6efa756d", response="7112ebf6630a249a788e898ee0123a26" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 260 v=0 o=root 1263250 1263251 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15984 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK435e0a94;received=159.65.48.104 From: ;tag=as6658275e To: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK435e0a94;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6658275e [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 (Checking To) --From tag as6658275e --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #23 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0a35965008fea95b4665220a212af999@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13395] channel.c: Channel 0x7f0c2004b3a0 'SIP/zvonobot-00000040' allocated [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13395] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13409] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:54] DEBUG[13404] channel.c: Channel 0x7f0c34096510 'SIP/zvonobot-00000041' allocated [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13404] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13395] res_stasis.c: calls_0: Subscribing to 213029 [Aug 18 10:33:54] DEBUG[13395] stasis/app.c: Channel '213029' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Outgoing Call for 79821117011 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13419] chan_sip.c: Audio is at 10922 [Aug 18 10:33:54] VERBOSE[13419] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13419] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13419] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13404] res_stasis.c: calls_0: Subscribing to 213030 [Aug 18 10:33:54] DEBUG[13404] stasis/app.c: Channel '213030' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Outgoing Call for 79821117010 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13420] chan_sip.c: Audio is at 16012 [Aug 18 10:33:54] VERBOSE[13420] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13420] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13420] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Initializing initreq for method INVITE - callid 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117010@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK008dedaf [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 3 [ 52]: From: ;tag=as6b014fb6 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 6 [ 60]: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13420] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117010@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK008dedaf Max-Forwards: 70 From: ;tag=as6b014fb6 To: Contact: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 26360097 26360097 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16012 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Aug 18 10:33:54] DEBUG[13420] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13404] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13404] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13395] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK008dedaf;received=159.65.48.104 From: ;tag=as6b014fb6 To: ;tag=as28bae283 Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0774fa7d" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK008dedaf;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6b014fb6 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as28bae283 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Initializing initreq for method INVITE - callid 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0774fa7d" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117011@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 (Checking To) --From tag as6b014fb6 --To-tag as28bae283 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6dc3346d [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 3 [ 52]: From: ;tag=as5c6f3360 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 6 [ 60]: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13419] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117011@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6dc3346d Max-Forwards: 70 From: ;tag=as5c6f3360 To: Contact: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1239511615 1239511615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10922 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Aug 18 10:33:54] DEBUG[13419] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '4005320d167483a22fe0427650e34c89@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:54] DEBUG[13395] http.c: HTTP closing session. Top level [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117010@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK008dedaf Max-Forwards: 70 From: ;tag=as6b014fb6 To: ;tag=as28bae283 Contact: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] VERBOSE[13420] dial.c: Called zvonobot/79821117010 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 16012 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13419] dial.c: Called zvonobot/79821117011 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: SIP call-id changed from '7c5d855836c6e2024acb04592e67a9ce@127.0.1.1:5060' to '6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060' [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117010@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bcbf932 Max-Forwards: 70 From: ;tag=as6b014fb6 To: Contact: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117010@178.62.121.41", nonce="0774fa7d", response="0fcdd94ef804ef5ad8e25b945bc60870" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 26360097 26360098 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16012 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13409] stasis.c: Creating topic. name: channel:213031, detail: [Aug 18 10:33:54] DEBUG[13409] stasis.c: Topic 'channel:213031': 0x7f0c7403fee0 created [Aug 18 10:33:54] DEBUG[13409] stasis.c: Creating topic. name: cache:140/channel:213031, detail: [Aug 18 10:33:54] DEBUG[13409] stasis.c: Topic 'cache:140/channel:213031': 0x7f0c74040950 created [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6dc3346d;received=159.65.48.104 From: ;tag=as5c6f3360 To: ;tag=as118b81fc Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52a05a14" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6dc3346d;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5c6f3360 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as118b81fc [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52a05a14" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 (Checking To) --From tag as5c6f3360 --To-tag as118b81fc [Aug 18 10:33:54] DEBUG[13407] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:54] DEBUG[13407] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13311] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117011@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6dc3346d Max-Forwards: 70 From: ;tag=as5c6f3360 To: ;tag=as118b81fc Contact: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 10922 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117011@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1995206f Max-Forwards: 70 From: ;tag=as5c6f3360 To: Contact: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117011@178.62.121.41", nonce="52a05a14", response="9b4f18aa165edda48636f60d363cc850" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1239511615 1239511616 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10922 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bcbf932;received=159.65.48.104 From: ;tag=as6b014fb6 To: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bcbf932;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6b014fb6 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4005320d167483a22fe0427650e34c89@159.65.48.104:5060 (Checking To) --From tag as6b014fb6 --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4005320d167483a22fe0427650e34c89@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[13403] channel.c: Channel 0x7f0c280bd370 'SIP/zvonobot-00000043' allocated [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13403] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13403] res_stasis.c: calls_0: Subscribing to 213026 [Aug 18 10:33:54] DEBUG[13403] stasis/app.c: Channel '213026' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Outgoing Call for 79821117014 [Aug 18 10:33:54] DEBUG[13403] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13403] http.c: HTTP closing session. Top level [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1995206f;received=159.65.48.104 From: ;tag=as5c6f3360 To: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1995206f;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5c6f3360 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13311] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 (Checking To) --From tag as5c6f3360 --To-tag [Aug 18 10:33:54] DEBUG[13311] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:54] DEBUG[13311] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #25 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13311] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:54] DEBUG[13311] channel.c: Channel Announcer/ARI-00000008;1 setting write format path: slin -> slin [Aug 18 10:33:54] DEBUG[13311] channel.c: Channel 0x2c32fc0 'Announcer/ARI-00000008;1' hanging up. Refs: 2 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13423] chan_sip.c: Audio is at 18660 [Aug 18 10:33:54] VERBOSE[13423] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13423] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13423] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Initializing initreq for method INVITE - callid 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117014@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c684877 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 3 [ 52]: From: ;tag=as452417ef [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 6 [ 60]: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13423] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117014@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c684877 Max-Forwards: 70 From: ;tag=as452417ef To: Contact: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 640369223 640369223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:33:54] DEBUG[13423] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] VERBOSE[13423] dial.c: Called zvonobot/79821117014 [Aug 18 10:33:54] DEBUG[13408] channel.c: Channel 0x7f0c4005cc40 'SIP/zvonobot-00000042' allocated [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13408] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[13408] res_stasis.c: calls_0: Subscribing to 213033 [Aug 18 10:33:54] DEBUG[13408] stasis/app.c: Channel '213033' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13408] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13408] http.c: HTTP closing session. Top level [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c684877;received=159.65.48.104 From: ;tag=as452417ef To: ;tag=as2e2d448b Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2abc798a" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[13406] channel.c: Channel 0x7f0c3c04f300 'SIP/zvonobot-00000044' allocated [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13406] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c684877;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as452417ef [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2e2d448b [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2abc798a" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 (Checking To) --From tag as452417ef --To-tag as2e2d448b [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117014@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c684877 Max-Forwards: 70 From: ;tag=as452417ef To: ;tag=as2e2d448b Contact: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Outgoing Call for 79821117007 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13406] res_stasis.c: calls_0: Subscribing to 213028 [Aug 18 10:33:54] VERBOSE[13425] chan_sip.c: Audio is at 19310 [Aug 18 10:33:54] VERBOSE[13425] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[13309] bridge_channel.c: Setting 0x2c3d160(Announcer/ARI-00000008;2) state from:0 to:1 [Aug 18 10:33:54] VERBOSE[13425] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13311] channel.c: Channel 0x2c32fc0 'Announcer/ARI-00000008;1' destroying [Aug 18 10:33:54] DEBUG[13406] stasis/app.c: Channel '213028' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13409] channel.c: Channel 0x7f0c7403c950 'SIP/zvonobot-00000045' allocated [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:54] DEBUG[13309] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pulling 0x2c3d160(Announcer/ARI-00000008;2) [Aug 18 10:33:54] VERBOSE[13425] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Initializing initreq for method INVITE - callid 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117007@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29ef9ec9 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 3 [ 52]: From: ;tag=as50732cd4 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 6 [ 60]: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13425] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117007@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29ef9ec9 Max-Forwards: 70 From: ;tag=as50732cd4 To: Contact: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1638458904 1638458904 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19310 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Aug 18 10:33:54] DEBUG[13425] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[13309] bridge_channel.c: Channel Announcer/ARI-00000008;2 left 'softmix' stasis-bridge [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13311] stasis.c: Destroying topic. name: cache:101/channel:1629282832.86, detail: [Aug 18 10:33:54] DEBUG[13311] stasis.c: Topic 'cache:101/channel:1629282832.86': 0x2c35b40 destroyed [Aug 18 10:33:54] DEBUG[13309] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x2c3d160(Announcer/ARI-00000008;2) is leaving softmix technology [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13406] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 18660 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[13311] stasis.c: Destroying topic. name: channel:1629282832.86, detail: [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13309] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59'. Checking compatability for channels 'SIP/zvonobot-0000001b' and 'Recorder/ARI-00000007;2' [Aug 18 10:33:54] DEBUG[13311] stasis.c: Topic 'channel:1629282832.86': 0x2c35110 destroyed [Aug 18 10:33:54] DEBUG[13309] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as channel 'SIP/zvonobot-0000001b' has features which prevent it [Aug 18 10:33:54] VERBOSE[13425] dial.c: Called zvonobot/79821117007 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Outgoing Call for 79821117012 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:54] DEBUG[13406] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:54] DEBUG[13409] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:54] VERBOSE[13426] chan_sip.c: Audio is at 16048 [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117014@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2 Max-Forwards: 70 From: ;tag=as452417ef To: Contact: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117014@178.62.121.41", nonce="2abc798a", response="09f70717293ccb201614aba438ff98d3" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 640369223 640369224 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17 [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13309] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:54] VERBOSE[13426] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: switching from softmix technology to simple_bridge [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology constructor [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c8c00b190(SIP/zvonobot-0000001b) to dummy bridge temporarily [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c9802d570(Recorder/ARI-00000007;2) to dummy bridge temporarily [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is leaving softmix technology (dummy) [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is leaving softmix technology (dummy) [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology stop [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is joining simple_bridge technology [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel Recorder/ARI-00000007;2 setting read format path: slin -> slin [Aug 18 10:33:54] VERBOSE[13426] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13426] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13409] res_stasis.c: calls_0: Subscribing to 213031 [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:54] DEBUG[13409] stasis/app.c: Channel '213031' is 1 interested in calls_0 [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining simple_bridge technology [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel Recorder/ARI-00000007;2 setting read format path: slin -> slin [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology start [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: deferring softmix technology destructor [Aug 18 10:33:54] DEBUG[13309] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: queueing action type:13 sub:1000 [Aug 18 10:33:54] DEBUG[13409] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:54] DEBUG[13409] http.c: HTTP closing session. Top level [Aug 18 10:33:54] DEBUG[13310] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: stopping mixing thread [Aug 18 10:33:54] DEBUG[13298] channel.c: Recorder/ARI-00000007;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel 0x2c3b150 'Announcer/ARI-00000008;2' hanging up. Refs: 2 [Aug 18 10:33:54] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:54] DEBUG[13294] channel.c: SIP/zvonobot-0000001b: Dropping redundant connected line update "" <>. [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Outgoing Call for 79821117009 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[13428] chan_sip.c: Audio is at 14410 [Aug 18 10:33:54] VERBOSE[13428] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] DEBUG[20534] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: Waiting for mixing thread to die. [Aug 18 10:33:54] VERBOSE[13428] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[13428] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29ef9ec9;received=159.65.48.104 From: ;tag=as50732cd4 To: ;tag=as11555cdf Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56dcd945" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Initializing initreq for method INVITE - callid 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117012@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Initializing initreq for method INVITE - callid 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a7097ad [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117009@178.62.121.41 SIP/2.0 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27c82536 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 3 [ 52]: From: ;tag=as0b9f5c0a [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 6 [ 60]: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13428] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117009@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27c82536 Max-Forwards: 70 From: ;tag=as0b9f5c0a To: Contact: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1668386055 1668386055 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14410 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Aug 18 10:33:54] DEBUG[13428] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] VERBOSE[13428] dial.c: Called zvonobot/79821117009 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29ef9ec9;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as50732cd4 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as11555cdf [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56dcd945" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 (Checking To) --From tag as50732cd4 --To-tag as11555cdf [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '10f580a044264908688c62534aa40882@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117007@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29ef9ec9 Max-Forwards: 70 From: ;tag=as50732cd4 To: ;tag=as11555cdf Contact: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 19310 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 3 [ 52]: From: ;tag=as22df0306 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 6 [ 60]: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117007@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30440186 Max-Forwards: 70 From: ;tag=as50732cd4 To: Contact: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117007@178.62.121.41", nonce="56dcd945", response="194c09c93a42202c29443131839d5d51" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1638458904 1638458905 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19310 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #26 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2;received=159.65.48.104 From: ;tag=as452417ef To: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as452417ef [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 (Checking To) --From tag as452417ef --To-tag [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #17 - INVITE (got response) [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:54 GMT [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:54] VERBOSE[13426] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117012@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a7097ad Max-Forwards: 70 From: ;tag=as22df0306 To: Contact: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 10847726 10847726 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27c82536;received=159.65.48.104 From: ;tag=as0b9f5c0a To: ;tag=as06e3d0f1 Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5881b904" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27c82536;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0b9f5c0a [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as06e3d0f1 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5881b904" [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 (Checking To) --From tag as0b9f5c0a --To-tag as06e3d0f1 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #24 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117009@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK27c82536 Max-Forwards: 70 From: ;tag=as0b9f5c0a To: ;tag=as06e3d0f1 Contact: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 14410 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117009@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ce8d885 Max-Forwards: 70 From: ;tag=as0b9f5c0a To: Contact: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117009@178.62.121.41", nonce="5881b904", response="8fa94a46b1a7733879dfd1199d043040" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1668386055 1668386056 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14410 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13426] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30440186;received=159.65.48.104 From: ;tag=as50732cd4 To: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30440186;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as50732cd4 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 (Checking To) --From tag as50732cd4 --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #26 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '10f580a044264908688c62534aa40882@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:54] DEBUG[13309] channel.c: Channel 0x2c3b150 'Announcer/ARI-00000008;2' destroying [Aug 18 10:33:54] DEBUG[13412] stasis.c: Creating topic. name: channel:1629282834.120, detail: [Aug 18 10:33:54] DEBUG[13412] stasis.c: Topic 'channel:1629282834.120': 0x7f0c7802c9a0 created [Aug 18 10:33:54] DEBUG[13412] stasis.c: Creating topic. name: cache:141/channel:1629282834.120, detail: [Aug 18 10:33:54] DEBUG[13412] stasis.c: Topic 'cache:141/channel:1629282834.120': 0x7f0c7804b7c0 created [Aug 18 10:33:54] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:54] DEBUG[13412] channel.c: Channel 0x7f0c780058a0 'Announcer/ARI-00000010;1' allocated [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ce8d885;received=159.65.48.104 From: ;tag=as0b9f5c0a To: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ce8d885;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0b9f5c0a [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[13412] stasis.c: Creating topic. name: channel:1629282834.121, detail: [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060 (Checking To) --From tag as0b9f5c0a --To-tag [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #28 - INVITE (got response) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6d114da20785bb564a6cdc72701a48e7@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:54] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:54] DEBUG[13412] stasis.c: Topic 'channel:1629282834.121': 0x7f0c78019b00 created [Aug 18 10:33:54] DEBUG[13412] stasis.c: Creating topic. name: cache:142/channel:1629282834.121, detail: [Aug 18 10:33:54] DEBUG[13412] stasis.c: Topic 'cache:142/channel:1629282834.121': 0x7f0c78019cb0 created [Aug 18 10:33:54] DEBUG[13412] channel.c: Channel 0x7f0c78050c90 'Announcer/ARI-00000010;2' allocated [Aug 18 10:33:54] DEBUG[13412] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:54] DEBUG[13412] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000010;1' [Aug 18 10:33:54] VERBOSE[13426] dial.c: Called zvonobot/79821117012 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a7097ad;received=159.65.48.104 From: ;tag=as22df0306 To: ;tag=as6bfca39f Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35308dad" Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:54] DEBUG[13431] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c7804b4e0(Announcer/ARI-00000010;2) is joining [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a7097ad;received=159.65.48.104 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22df0306 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6bfca39f [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35308dad" [Aug 18 10:33:54] DEBUG[13309] stasis.c: Destroying topic. name: cache:102/channel:1629282832.87, detail: [Aug 18 10:33:54] DEBUG[13309] stasis.c: Topic 'cache:102/channel:1629282832.87': 0x2c3e010 destroyed [Aug 18 10:33:54] DEBUG[13309] stasis.c: Destroying topic. name: channel:1629282832.87, detail: [Aug 18 10:33:54] DEBUG[13309] stasis.c: Topic 'channel:1629282832.87': 0x2c3d590 destroyed [Aug 18 10:33:54] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:54] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:54] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 (Checking To) --From tag as22df0306 --To-tag as6bfca39f [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #24 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Stopping retransmission on '72b609e44febd02a23174a4d311879ff@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:54] DEBUG[13431] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pushing 0x7f0c7804b4e0(Announcer/ARI-00000010;2) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117012@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a7097ad Max-Forwards: 70 From: ;tag=as22df0306 To: ;tag=as6bfca39f Contact: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Audio is at 16048 [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117012@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c14d3a0 Max-Forwards: 70 From: ;tag=as22df0306 To: Contact: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117012@178.62.121.41", nonce="35308dad", response="e7d5722bc506ea4b02415910d505a7d3" Date: Wed, 18 Aug 2021 10:33:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 10847726 10847727 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:33:54] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:54] DEBUG[13431] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:54] VERBOSE[13431] bridge_channel.c: Channel Announcer/ARI-00000010;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:54] DEBUG[13431] bridge.c: Chose bridge technology softmix [Aug 18 10:33:54] VERBOSE[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: switching from simple_bridge technology to softmix [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology constructor [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c8c00b190(SIP/zvonobot-0000001b) to dummy bridge temporarily [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c9802d570(Recorder/ARI-00000007;2) to dummy bridge temporarily [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is leaving simple_bridge technology (dummy) [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology stop [Aug 18 10:33:54] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c14d3a0;received=159.65.48.104 From: ;tag=as22df0306 To: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c7804b4e0(Announcer/ARI-00000010;2) is joining softmix technology [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: Announcer/ARI-00000010;2: [Aug 18 10:33:54] DEBUG[13431] channel.c: Channel Announcer/ARI-00000010;2 setting write format path: slin -> slin [Aug 18 10:33:54] DEBUG[13431] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:54] DEBUG[13431] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: Announcer/ARI-00000010;2: [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: Announcer/ARI-00000010;2: Not in SFU mode [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is joining softmix technology [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: SIP/zvonobot-0000001b: [Aug 18 10:33:54] DEBUG[13431] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:54] DEBUG[13431] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: SIP/zvonobot-0000001b: [Aug 18 10:33:54] DEBUG[13431] bridge_softmix.c: SIP/zvonobot-0000001b: Not in SFU mode [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:54] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c14d3a0;received=159.65.48.104 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22df0306 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:33:55] DEBUG[13431] bridge_softmix.c: Recorder/ARI-00000007;2: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 (Checking To) --From tag as22df0306 --To-tag [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #21 - INVITE (got response) [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '72b609e44febd02a23174a4d311879ff@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:55] DEBUG[13431] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13431] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13431] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13431] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13431] bridge_softmix.c: Recorder/ARI-00000007;2: [Aug 18 10:33:55] DEBUG[13431] bridge_softmix.c: Recorder/ARI-00000007;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology start [Aug 18 10:33:55] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology destructor [Aug 18 10:33:55] DEBUG[13412] res_stasis_playback.c: 1629282834.120: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:55] DEBUG[13412] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13433] channel.c: Channel Announcer/ARI-00000010;1 setting write format path: gsm -> slin [Aug 18 10:33:55] DEBUG[13412] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13432] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: starting mixing thread [Aug 18 10:33:55] DEBUG[13433] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13294] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTP audio difference is 1168, ms is 166 [Aug 18 10:33:55] VERBOSE[13433] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7cfa6980;received=159.65.48.104 From: ;tag=as0d63cc42 To: ;tag=as6847ab41 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 2010043336 2010043336 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10912 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7cfa6980;received=159.65.48.104 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0d63cc42 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6847ab41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 2010043336 2010043336 IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10912 RTP/AVP 0 8 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: = Looking for Call ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 (Checking To) --From tag as0d63cc42 --To-tag as6847ab41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Stopping retransmission on '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Got SDP version 2010043336 and unique parts [root 2010043336 IN IP4 178.62.121.41] [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 2010043336 2010043336 IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE set role failed; no ice instance [Aug 18 10:33:55] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP setting address on RTP instance [Aug 18 10:33:55] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c8c013990 -- Strict RTP learning after remote address set to: 178.62.121.41:10912 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10912 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00cd7e8) from 0x7f0c147e2330 to 0x7f0c8c00f9b8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00cd838) from 0x7f0c147e2330 to 0x7f0c8c00f9b8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0092008) from 0x7f0c147e2330 to 0x7f0c8c00f9b8 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP ignoring duplicate property [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000020 setting read format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000020 setting write format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS - ast_rtp_activate rtp=0x7f0c8c013990 - setup and perform DTLS' [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c013990) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c013990) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:55] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Strict routing enforced for session 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117045@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7d74ea27 Max-Forwards: 70 From: ;tag=as0d63cc42 To: ;tag=as6847ab41 Contact: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session timer started: 20 - 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 1768000ms [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:55] VERBOSE[13114] dial.c: SIP/zvonobot-00000020 answered [Aug 18 10:33:55] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:55] VERBOSE[13114] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000020 [Aug 18 10:33:55] DEBUG[13114] stasis/app.c: Channel '212995' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:55] DEBUG[13434] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13434] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13434] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13434] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13434] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13434] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13434] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13434] stasis.c: Creating topic. name: bridge:0aaea81d-67a8-499e-9e08-2fb745e40804, detail: [Aug 18 10:33:55] DEBUG[13434] stasis.c: Topic 'bridge:0aaea81d-67a8-499e-9e08-2fb745e40804': 0x7f0cb00d0d80 created [Aug 18 10:33:55] DEBUG[13434] stasis.c: Creating topic. name: cache:143/bridge:0aaea81d-67a8-499e-9e08-2fb745e40804, detail: [Aug 18 10:33:55] DEBUG[13434] stasis.c: Topic 'cache:143/bridge:0aaea81d-67a8-499e-9e08-2fb745e40804': 0x7f0cb00c9240 created [Aug 18 10:33:55] DEBUG[13434] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13434] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13434] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13434] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13434] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13434] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13434] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13434] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13434] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13435] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13435] http.c: HTTP Request URI is /ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/addChannel?channel=212995 [Aug 18 10:33:55] DEBUG[13435] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13435] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13435] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/addChannel] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13435] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Finding handler for bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/addChannel [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Finding handler for 0aaea81d-67a8-499e-9e08-2fb745e40804 [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13435] res_ari.c: No explicit handler found for 0aaea81d-67a8-499e-9e08-2fb745e40804. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Finding handler for addChannel [Aug 18 10:33:55] DEBUG[13435] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:55] DEBUG[13435] stasis/control.c: 212995: Sending channel add_to_bridge command [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f2da01;received=159.65.48.104 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 512104757 512104757 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16938 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f2da01;received=159.65.48.104 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a1fc7ed [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3f55a57d [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 512104757 512104757 IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16938 RTP/AVP 0 8 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: = Looking for Call ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 (Checking To) --From tag as3a1fc7ed --To-tag as3f55a57d [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Stopping retransmission on '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Got SDP version 512104757 and unique parts [root 512104757 IN IP4 178.62.121.41] [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 512104757 512104757 IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE set role failed; no ice instance [Aug 18 10:33:55] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) RTCP setting address on RTP instance [Aug 18 10:33:55] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c9800ef10 -- Strict RTP learning after remote address set to: 178.62.121.41:16938 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:16938 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00cbc38) from 0x7f0c147e2330 to 0x7f0c9800b198 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0097388) from 0x7f0c147e2330 to 0x7f0c9800b198 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00d39e8) from 0x7f0c147e2330 to 0x7f0c9800b198 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) RTCP ignoring duplicate property [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000023 setting read format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000023 setting write format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS - ast_rtp_activate rtp=0x7f0c9800ef10 - setup and perform DTLS' [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800ef10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800ef10) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:55] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Strict routing enforced for session 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01ed668f Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Contact: Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session timer started: 30 - 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 1768000ms [Aug 18 10:33:55] VERBOSE[13129] dial.c: SIP/zvonobot-00000023 answered [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:55] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:55] VERBOSE[13129] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000023 [Aug 18 10:33:55] DEBUG[13129] stasis/app.c: Channel '212999' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:55] DEBUG[13436] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13436] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13436] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13436] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13436] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13436] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13436] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13436] stasis.c: Creating topic. name: bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162, detail: [Aug 18 10:33:55] DEBUG[13436] stasis.c: Topic 'bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162': 0x7f0cb404aff0 created [Aug 18 10:33:55] DEBUG[13436] stasis.c: Creating topic. name: cache:144/bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162, detail: [Aug 18 10:33:55] DEBUG[13436] stasis.c: Topic 'cache:144/bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162': 0x7f0cb4045cd0 created [Aug 18 10:33:55] DEBUG[13436] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13436] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13436] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13436] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13436] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13436] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13436] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13436] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13436] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13437] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13437] http.c: HTTP Request URI is /ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/addChannel?channel=212999 [Aug 18 10:33:55] DEBUG[13437] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13437] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13437] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/addChannel] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13437] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Finding handler for bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/addChannel [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Finding handler for 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13437] res_ari.c: No explicit handler found for 3d1f9573-48d1-48a0-bcdd-9d7f09555162. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Finding handler for addChannel [Aug 18 10:33:55] DEBUG[13437] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:55] DEBUG[13437] stasis/control.c: 212999: Sending channel add_to_bridge command [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK12ed07ac;received=159.65.48.104 From: ;tag=as510b84fe To: ;tag=as6657c8e8 Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 137985340 137985340 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13636 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK12ed07ac;received=159.65.48.104 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as510b84fe [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6657c8e8 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 137985340 137985340 IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13636 RTP/AVP 0 8 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: = Looking for Call ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 (Checking To) --From tag as510b84fe --To-tag as6657c8e8 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Stopping retransmission on '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Got SDP version 137985340 and unique parts [root 137985340 IN IP4 178.62.121.41] [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 137985340 137985340 IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) ICE set role failed; no ice instance [Aug 18 10:33:55] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) RTCP setting address on RTP instance [Aug 18 10:33:55] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c94023d30 -- Strict RTP learning after remote address set to: 178.62.121.41:13636 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:13636 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00b10a8) from 0x7f0c147e2330 to 0x7f0c940227e8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00d3e48) from 0x7f0c147e2330 to 0x7f0c940227e8 [Aug 18 10:33:55] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0066818) from 0x7f0c147e2330 to 0x7f0c940227e8 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) RTCP ignoring duplicate property [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002e setting read format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002e setting write format path: alaw -> alaw [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS - ast_rtp_activate rtp=0x7f0c94023d30 - setup and perform DTLS' [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94023d30) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94023d30) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:55] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Strict routing enforced for session 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:55] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:55] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:55] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117028@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b14a6db Max-Forwards: 70 From: ;tag=as510b84fe To: ;tag=as6657c8e8 Contact: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:55] DEBUG[20585] chan_sip.c: Session timer started: 22 - 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 1768000ms [Aug 18 10:33:55] VERBOSE[13195] dial.c: SIP/zvonobot-0000002e answered [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:55] VERBOSE[13195] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000002e [Aug 18 10:33:55] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:55] DEBUG[13195] stasis/app.c: Channel '213012' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:55] DEBUG[13439] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13439] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13439] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13439] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13439] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13439] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13439] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13439] stasis.c: Creating topic. name: bridge:e2e70698-2279-429d-a48c-2fe9dd817267, detail: [Aug 18 10:33:55] DEBUG[13439] stasis.c: Topic 'bridge:e2e70698-2279-429d-a48c-2fe9dd817267': 0x7f0c100177b0 created [Aug 18 10:33:55] DEBUG[13439] stasis.c: Creating topic. name: cache:145/bridge:e2e70698-2279-429d-a48c-2fe9dd817267, detail: [Aug 18 10:33:55] DEBUG[13439] stasis.c: Topic 'cache:145/bridge:e2e70698-2279-429d-a48c-2fe9dd817267': 0x7f0c10065e40 created [Aug 18 10:33:55] DEBUG[13439] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13439] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13439] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13439] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13439] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13439] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13439] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13439] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13439] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13440] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13440] http.c: HTTP Request URI is /ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/addChannel?channel=213012 [Aug 18 10:33:55] DEBUG[13440] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13440] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13440] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/addChannel] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13440] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Finding handler for bridges/e2e70698-2279-429d-a48c-2fe9dd817267/addChannel [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Finding handler for e2e70698-2279-429d-a48c-2fe9dd817267 [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13440] res_ari.c: No explicit handler found for e2e70698-2279-429d-a48c-2fe9dd817267. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Finding handler for addChannel [Aug 18 10:33:55] DEBUG[13440] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:55] DEBUG[13440] stasis/control.c: 213012: Sending channel add_to_bridge command [Aug 18 10:33:55] VERBOSE[13208] res_rtp_asterisk.c: 0x7f0c100223b0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:10788 [Aug 18 10:33:55] DEBUG[13114] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000020 [Aug 18 10:33:55] DEBUG[13114] stasis/control.c: 212995: Adding to bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 [Aug 18 10:33:55] DEBUG[13114] stasis/app.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13441] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is joining [Aug 18 10:33:55] DEBUG[13441] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pushing 0x7f0c88050c90(SIP/zvonobot-00000020) [Aug 18 10:33:55] VERBOSE[13441] bridge_channel.c: Channel SIP/zvonobot-00000020 joined 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:33:55] DEBUG[13441] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 is already using the new technology. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP changing ssrc from 770552814 to 1690663494 due to a source change [Aug 18 10:33:55] DEBUG[13435] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:55] DEBUG[13435] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13442] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13114] stasis/app.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[13442] http.c: HTTP Request URI is /ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/record?name=212995_uVwfgYUYeUZPIEIoeKlVpacDKSnRhOIB&format=wav [Aug 18 10:33:55] DEBUG[13442] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/record] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13442] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/record] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13442] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/record] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13442] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Finding handler for bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Finding handler for 0aaea81d-67a8-499e-9e08-2fb745e40804 [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13442] res_ari.c: No explicit handler found for 0aaea81d-67a8-499e-9e08-2fb745e40804. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Finding handler for record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:55] DEBUG[13442] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:55] DEBUG[13442] stasis.c: Creating topic. name: channel:1629282835.122, detail: [Aug 18 10:33:55] DEBUG[13442] stasis.c: Topic 'channel:1629282835.122': 0x7f0c1807fc30 created [Aug 18 10:33:55] DEBUG[13442] stasis.c: Creating topic. name: cache:146/channel:1629282835.122, detail: [Aug 18 10:33:55] DEBUG[13442] stasis.c: Topic 'cache:146/channel:1629282835.122': 0x7f0c1800c330 created [Aug 18 10:33:55] DEBUG[13442] channel.c: Channel 0x7f0c180acf90 'Recorder/ARI-00000011;1' allocated [Aug 18 10:33:55] DEBUG[13442] stasis.c: Creating topic. name: channel:1629282835.123, detail: [Aug 18 10:33:55] DEBUG[13442] stasis.c: Topic 'channel:1629282835.123': 0x7f0c18093e40 created [Aug 18 10:33:55] DEBUG[13442] stasis.c: Creating topic. name: cache:147/channel:1629282835.123, detail: [Aug 18 10:33:55] DEBUG[13442] stasis.c: Topic 'cache:147/channel:1629282835.123': 0x7f0c1800e6f0 created [Aug 18 10:33:55] DEBUG[13442] channel.c: Channel 0x7f0c18087070 'Recorder/ARI-00000011;2' allocated [Aug 18 10:33:55] DEBUG[13442] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] DEBUG[13443] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining [Aug 18 10:33:55] DEBUG[13443] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pushing 0x7f0c180a6470(Recorder/ARI-00000011;2) [Aug 18 10:33:55] DEBUG[13443] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] VERBOSE[13443] bridge_channel.c: Channel Recorder/ARI-00000011;2 joined 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:33:55] DEBUG[13443] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804'. Checking compatability for channels 'SIP/zvonobot-00000020' and 'Recorder/ARI-00000011;2' [Aug 18 10:33:55] DEBUG[13443] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as could not get details [Aug 18 10:33:55] DEBUG[13443] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13443] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13443] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13443] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13443] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 is already using the new technology. [Aug 18 10:33:55] DEBUG[13443] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13443] channel.c: Channel Recorder/ARI-00000011;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[13443] channel.c: Channel SIP/zvonobot-00000020 setting write format path: slin -> alaw [Aug 18 10:33:55] DEBUG[13443] channel.c: Channel SIP/zvonobot-00000020 setting read format path: alaw -> slin [Aug 18 10:33:55] DEBUG[13443] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13442] res_stasis_recording.c: 1629282835.122: Sending record(212995_uVwfgYUYeUZPIEIoeKlVpacDKSnRhOIB.wav) command [Aug 18 10:33:55] DEBUG[13442] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13444] app.c: play_and_record: , /var/spool/asterisk/recording/212995_uVwfgYUYeUZPIEIoeKlVpacDKSnRhOIB, 'wav' [Aug 18 10:33:55] DEBUG[13444] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:55] VERBOSE[13444] app.c: x=0, open writing: /var/spool/asterisk/recording/212995_uVwfgYUYeUZPIEIoeKlVpacDKSnRhOIB format: wav, 0x7f0c2004acc0 [Aug 18 10:33:55] DEBUG[13442] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13445] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13445] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13445] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13445] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13445] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13445] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13445] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13445] stasis.c: Creating topic. name: bridge:d177377e-a80b-4ad9-826a-cece7d5abce5, detail: [Aug 18 10:33:55] DEBUG[13445] stasis.c: Topic 'bridge:d177377e-a80b-4ad9-826a-cece7d5abce5': 0x7f0c2c055b20 created [Aug 18 10:33:55] DEBUG[13445] stasis.c: Creating topic. name: cache:148/bridge:d177377e-a80b-4ad9-826a-cece7d5abce5, detail: [Aug 18 10:33:55] DEBUG[13445] stasis.c: Topic 'cache:148/bridge:d177377e-a80b-4ad9-826a-cece7d5abce5': 0x7f0c2c0526f0 created [Aug 18 10:33:55] DEBUG[13445] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13445] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13445] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13445] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13445] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13445] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13445] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13445] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13445] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13446] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13446] http.c: HTTP Request URI is /ari/channels/212995/snoop?app=calls_0&spy=in [Aug 18 10:33:55] DEBUG[13446] http.c: match request [ari/channels/212995/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13446] http.c: match request [ari/channels/212995/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13446] http.c: match request [ari/channels/212995/snoop] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13446] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Finding handler for channels/212995/snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Finding handler for 212995 [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channels create: Didn't match 212995 [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channels externalMedia: Didn't match 212995 [Aug 18 10:33:55] DEBUG[13446] res_ari.c: No explicit handler found for 212995. Using wildcard channelId. [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Finding handler for snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:55] DEBUG[13446] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:55] DEBUG[13446] stasis.c: Creating topic. name: channel:1629282835.124, detail: [Aug 18 10:33:55] DEBUG[13446] stasis.c: Topic 'channel:1629282835.124': 0x7f0c280a2df0 created [Aug 18 10:33:55] DEBUG[13446] stasis.c: Creating topic. name: cache:149/channel:1629282835.124, detail: [Aug 18 10:33:55] DEBUG[13446] stasis.c: Topic 'cache:149/channel:1629282835.124': 0x7f0c280b3ac0 created [Aug 18 10:33:55] DEBUG[13446] channel.c: Channel 0x7f0c280c6fb0 'Snoop/212995-00000007' allocated [Aug 18 10:33:55] DEBUG[13441] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804'. Checking compatability for channels 'SIP/zvonobot-00000020' and 'Recorder/ARI-00000011;2' [Aug 18 10:33:55] DEBUG[13441] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as channel 'SIP/zvonobot-00000020' has features which prevent it [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13441] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 is already using the new technology. [Aug 18 10:33:55] DEBUG[13446] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13446] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13447] stasis/app.c: Channel '1629282835.124' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13453] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13453] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212995&app=calls_0&format=slin16&external_host=127.0.0.1%3A50220 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:55] DEBUG[13450] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13450] http.c: HTTP Request URI is /ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play?media=sound%3Asilence%2F2 [Aug 18 10:33:55] DEBUG[13450] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13450] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13450] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13450] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Finding handler for bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play [Aug 18 10:33:55] DEBUG[13453] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13453] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13453] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13453] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Finding handler for 0aaea81d-67a8-499e-9e08-2fb745e40804 [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: No explicit handler found for 0aaea81d-67a8-499e-9e08-2fb745e40804. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:55] DEBUG[13450] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13450] stasis.c: Creating topic. name: channel:1629282835.125, detail: [Aug 18 10:33:55] DEBUG[13453] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:55] DEBUG[13450] stasis.c: Topic 'channel:1629282835.125': 0x7f0c30071040 created [Aug 18 10:33:55] DEBUG[13450] stasis.c: Creating topic. name: cache:150/channel:1629282835.125, detail: [Aug 18 10:33:55] DEBUG[13453] netsock2.c: Splitting '127.0.0.1:50220' into... [Aug 18 10:33:55] DEBUG[13450] stasis.c: Topic 'cache:150/channel:1629282835.125': 0x7f0c3007c870 created [Aug 18 10:33:55] DEBUG[13453] netsock2.c: ...host '127.0.0.1' and port '50220'. [Aug 18 10:33:55] DEBUG[13453] netsock2.c: Splitting '127.0.0.1:50220' into... [Aug 18 10:33:55] DEBUG[13450] channel.c: Channel 0x7f0c3009e0e0 'Announcer/ARI-00000012;1' allocated [Aug 18 10:33:55] DEBUG[13453] netsock2.c: ...host '127.0.0.1' and port '50220'. [Aug 18 10:33:55] DEBUG[13453] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:55] DEBUG[13453] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c052f40' [Aug 18 10:33:55] DEBUG[13450] stasis.c: Creating topic. name: channel:1629282835.126, detail: [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP allocated port 11092 [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) ICE creating session 127.0.0.1:11092 (11092) [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) ICE create [Aug 18 10:33:55] DEBUG[13450] stasis.c: Topic 'channel:1629282835.126': 0x7f0c3007c7b0 created [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) ICE add system candidates [Aug 18 10:33:55] DEBUG[13450] stasis.c: Creating topic. name: cache:151/channel:1629282835.126, detail: [Aug 18 10:33:55] DEBUG[13450] stasis.c: Topic 'cache:151/channel:1629282835.126': 0x7f0c30071ce0 created [Aug 18 10:33:55] DEBUG[13453] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:55] DEBUG[13453] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) ICE add candidate: 159.65.48.104:11092, 2130706431 [Aug 18 10:33:55] DEBUG[13453] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:55] DEBUG[13453] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:55] DEBUG[13453] res_rtp_asterisk.c: (0x7f0c3c052f40) ICE add candidate: 10.131.0.10:11092, 2130706431 [Aug 18 10:33:55] DEBUG[13453] rtp_engine.c: RTP instance '0x7f0c3c052f40' is setup and ready to go [Aug 18 10:33:55] DEBUG[13453] stasis.c: Creating topic. name: channel:robot_212995, detail: [Aug 18 10:33:55] DEBUG[13453] stasis.c: Topic 'channel:robot_212995': 0x7f0c3c061bd0 created [Aug 18 10:33:55] DEBUG[13453] stasis.c: Creating topic. name: cache:152/channel:robot_212995, detail: [Aug 18 10:33:55] DEBUG[13453] stasis.c: Topic 'cache:152/channel:robot_212995': 0x7f0c3c0626c0 created [Aug 18 10:33:55] DEBUG[13450] channel.c: Channel 0x7f0c300a36a0 'Announcer/ARI-00000012;2' allocated [Aug 18 10:33:55] DEBUG[13129] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000023 [Aug 18 10:33:55] DEBUG[13129] stasis/control.c: 212999: Adding to bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:33:55] DEBUG[13129] stasis/app.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13454] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining [Aug 18 10:33:55] DEBUG[13454] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pushing 0x7f0ca0053060(SIP/zvonobot-00000023) [Aug 18 10:33:55] DEBUG[13450] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] VERBOSE[13454] bridge_channel.c: Channel SIP/zvonobot-00000023 joined 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:33:55] DEBUG[13450] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000012;1' [Aug 18 10:33:55] DEBUG[13453] channel.c: Channel 0x7f0c3c05fae0 'UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40' allocated [Aug 18 10:33:55] DEBUG[13455] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c300a4c90(Announcer/ARI-00000012;2) is joining [Aug 18 10:33:55] DEBUG[13453] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:55] VERBOSE[13453] res_rtp_asterisk.c: 0x7f0c3c05b0c0 -- Strict RTP learning after remote address set to: 127.0.0.1:50220 [Aug 18 10:33:55] DEBUG[13455] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pushing 0x7f0c300a4c90(Announcer/ARI-00000012;2) [Aug 18 10:33:55] DEBUG[13455] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] VERBOSE[13455] bridge_channel.c: Channel Announcer/ARI-00000012;2 joined 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13455] bridge.c: Chose bridge technology softmix [Aug 18 10:33:55] VERBOSE[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: switching from simple_bridge technology to softmix [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology constructor [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c88050c90(SIP/zvonobot-00000020) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c180a6470(Recorder/ARI-00000011;2) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology stop [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c300a4c90(Announcer/ARI-00000012;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Announcer/ARI-00000012;2: [Aug 18 10:33:55] DEBUG[13453] res_stasis.c: calls_0: Subscribing to robot_212995 [Aug 18 10:33:55] DEBUG[13454] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13455] channel.c: Channel Announcer/ARI-00000012;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Announcer/ARI-00000012;2: [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Announcer/ARI-00000012;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is joining softmix technology [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: SIP/zvonobot-00000020: [Aug 18 10:33:55] DEBUG[13453] stasis/app.c: Channel 'robot_212995' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13453] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13454] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 is already using the new technology. [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13453] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: SIP/zvonobot-00000020: [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: SIP/zvonobot-00000020: Not in SFU mode [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:55] DEBUG[13129] stasis/app.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' is 2 interested in calls_0 [Aug 18 10:33:55] VERBOSE[13457] dial.c: Called 127.0.0.1:50220 [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50220 [Aug 18 10:33:55] VERBOSE[13457] dial.c: UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 answered [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13437] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:55] DEBUG[13437] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13458] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13458] http.c: HTTP Request URI is /ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/record?name=212999_qRuswmdCeCDHQrZUypTzhhPimeblcjmR&format=wav [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Recorder/ARI-00000011;2: [Aug 18 10:33:55] DEBUG[13458] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/record] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP changing ssrc from 1163544665 to 1741846571 due to a source change [Aug 18 10:33:55] DEBUG[13455] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:33:55] VERBOSE[13457] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 [Aug 18 10:33:55] DEBUG[13195] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000002e [Aug 18 10:33:55] DEBUG[13195] stasis/control.c: 213012: Adding to bridge e2e70698-2279-429d-a48c-2fe9dd817267 [Aug 18 10:33:55] DEBUG[13195] stasis/app.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13458] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/record] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13458] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/record] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13455] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Recorder/ARI-00000011;2: [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:55] DEBUG[13455] bridge_softmix.c: Recorder/ARI-00000011;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology start [Aug 18 10:33:55] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology destructor [Aug 18 10:33:55] DEBUG[13457] stasis/app.c: Channel 'robot_212995' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[13450] res_stasis_playback.c: 1629282835.125: Sending play(sound:silence/2) command [Aug 18 10:33:55] DEBUG[13456] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: starting mixing thread [Aug 18 10:33:55] DEBUG[13460] channel.c: Channel Announcer/ARI-00000012;1 setting write format path: gsm -> slin [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50220 - state 2 (In use) [Aug 18 10:33:55] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50220, detail: [Aug 18 10:33:55] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50220': 0x7f0c84067fd0 created [Aug 18 10:33:55] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50220' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:55] DEBUG[13450] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13450] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13459] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is joining [Aug 18 10:33:55] DEBUG[13458] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:33:55] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP ooh, format changed from none to alaw [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Finding handler for bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/record [Aug 18 10:33:55] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP starting transmission [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Finding handler for 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13458] res_ari.c: No explicit handler found for 3d1f9573-48d1-48a0-bcdd-9d7f09555162. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Finding handler for record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:55] DEBUG[13458] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:55] DEBUG[13458] stasis.c: Creating topic. name: channel:1629282835.128, detail: [Aug 18 10:33:55] DEBUG[13458] stasis.c: Topic 'channel:1629282835.128': 0x7f0c7c01ae00 created [Aug 18 10:33:55] DEBUG[13458] stasis.c: Creating topic. name: cache:153/channel:1629282835.128, detail: [Aug 18 10:33:55] DEBUG[13458] stasis.c: Topic 'cache:153/channel:1629282835.128': 0x7f0c7c010d50 created [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:55] DEBUG[13458] channel.c: Channel 0x7f0c7c00f100 'Recorder/ARI-00000013;1' allocated [Aug 18 10:33:55] DEBUG[13458] stasis.c: Creating topic. name: channel:1629282835.129, detail: [Aug 18 10:33:55] DEBUG[13458] stasis.c: Topic 'channel:1629282835.129': 0x7f0c7c047e00 created [Aug 18 10:33:55] DEBUG[13458] stasis.c: Creating topic. name: cache:154/channel:1629282835.129, detail: [Aug 18 10:33:55] DEBUG[13458] stasis.c: Topic 'cache:154/channel:1629282835.129': 0x7f0c7c016ec0 created [Aug 18 10:33:55] DEBUG[13458] channel.c: Channel 0x7f0c7c017b90 'Recorder/ARI-00000013;2' allocated [Aug 18 10:33:55] DEBUG[13458] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] DEBUG[13460] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:55] VERBOSE[13441] res_rtp_asterisk.c: 0x7f0c8c013990 -- Strict RTP switching to RTP target address 178.62.121.41:10912 as source [Aug 18 10:33:55] DEBUG[13459] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pushing 0x7f0c9006b170(SIP/zvonobot-0000002e) [Aug 18 10:33:55] DEBUG[13462] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining [Aug 18 10:33:55] VERBOSE[13460] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:55] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:33:55] DEBUG[13461] http.c: HTTP opening session. Top level [Aug 18 10:33:55] VERBOSE[13459] bridge_channel.c: Channel SIP/zvonobot-0000002e joined 'simple_bridge' stasis-bridge [Aug 18 10:33:55] DEBUG[13459] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13462] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pushing 0x7f0c7c018d60(Recorder/ARI-00000013;2) [Aug 18 10:33:55] DEBUG[13461] http.c: HTTP Request URI is /ari/bridges/d177377e-a80b-4ad9-826a-cece7d5abce5/addChannel?channel=1629282835.124%2Crobot_212995 [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13462] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] VERBOSE[13462] bridge_channel.c: Channel Recorder/ARI-00000013;2 joined 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13461] http.c: match request [ari/bridges/d177377e-a80b-4ad9-826a-cece7d5abce5/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13459] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267 is already using the new technology. [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13461] http.c: match request [ari/bridges/d177377e-a80b-4ad9-826a-cece7d5abce5/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP changing ssrc from 1693060834 to 2068879853 due to a source change [Aug 18 10:33:55] DEBUG[13195] stasis/app.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[13440] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:55] DEBUG[13462] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162'. Checking compatability for channels 'SIP/zvonobot-00000023' and 'Recorder/ARI-00000013;2' [Aug 18 10:33:55] DEBUG[13462] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as could not get details [Aug 18 10:33:55] DEBUG[13440] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13462] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13463] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13462] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13462] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13462] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13463] http.c: HTTP Request URI is /ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/record?name=213012_IbmNZTtYydMrsTzVrzfNFHIyzKWXigCw&format=wav [Aug 18 10:33:55] DEBUG[13462] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 is already using the new technology. [Aug 18 10:33:55] DEBUG[13463] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/record] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13463] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/record] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13463] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/record] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13463] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13462] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13461] http.c: match request [ari/bridges/d177377e-a80b-4ad9-826a-cece7d5abce5/addChannel] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13461] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13462] channel.c: Channel Recorder/ARI-00000013;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Finding handler for bridges/d177377e-a80b-4ad9-826a-cece7d5abce5/addChannel [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Finding handler for bridges/e2e70698-2279-429d-a48c-2fe9dd817267/record [Aug 18 10:33:55] DEBUG[13462] channel.c: Channel SIP/zvonobot-00000023 setting write format path: slin -> alaw [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13462] channel.c: Channel SIP/zvonobot-00000023 setting read format path: alaw -> slin [Aug 18 10:33:55] DEBUG[13462] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Finding handler for e2e70698-2279-429d-a48c-2fe9dd817267 [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: No explicit handler found for e2e70698-2279-429d-a48c-2fe9dd817267. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Finding handler for record [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Finding handler for d177377e-a80b-4ad9-826a-cece7d5abce5 [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13461] res_ari.c: No explicit handler found for d177377e-a80b-4ad9-826a-cece7d5abce5. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:55] DEBUG[13463] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Finding handler for addChannel [Aug 18 10:33:55] DEBUG[13461] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:55] DEBUG[13463] stasis.c: Creating topic. name: channel:1629282835.130, detail: [Aug 18 10:33:55] DEBUG[13461] stasis/control.c: 1629282835.124: Sending channel add_to_bridge command [Aug 18 10:33:55] DEBUG[13463] stasis.c: Topic 'channel:1629282835.130': 0x7f0c88080d50 created [Aug 18 10:33:55] DEBUG[13463] stasis.c: Creating topic. name: cache:155/channel:1629282835.130, detail: [Aug 18 10:33:55] DEBUG[13464] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13464] http.c: HTTP Request URI is /ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:55] DEBUG[13463] stasis.c: Topic 'cache:155/channel:1629282835.130': 0x7f0c88080f20 created [Aug 18 10:33:55] DEBUG[13464] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13464] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13464] http.c: match request [ari/bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13464] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Finding handler for bridges/0ec77f0c-7a86-4072-a1c4-e42f5256208c/play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13463] channel.c: Channel 0x7f0c88078fc0 'Recorder/ARI-00000014;1' allocated [Aug 18 10:33:55] DEBUG[13458] res_stasis_recording.c: 1629282835.128: Sending record(212999_qRuswmdCeCDHQrZUypTzhhPimeblcjmR.wav) command [Aug 18 10:33:55] DEBUG[13463] stasis.c: Creating topic. name: channel:1629282835.131, detail: [Aug 18 10:33:55] DEBUG[13463] stasis.c: Topic 'channel:1629282835.131': 0x7f0c880747d0 created [Aug 18 10:33:55] DEBUG[13463] stasis.c: Creating topic. name: cache:156/channel:1629282835.131, detail: [Aug 18 10:33:55] DEBUG[13463] stasis.c: Topic 'cache:156/channel:1629282835.131': 0x7f0c88087db0 created [Aug 18 10:33:55] DEBUG[13447] bridge_roles.c: Roles did not exist on channel Snoop/212995-00000007 [Aug 18 10:33:55] DEBUG[13447] stasis/control.c: 1629282835.124: Adding to bridge d177377e-a80b-4ad9-826a-cece7d5abce5 [Aug 18 10:33:55] DEBUG[13447] stasis/app.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13458] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13458] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13466] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c34027b30(Snoop/212995-00000007) is joining [Aug 18 10:33:55] DEBUG[13466] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: pushing 0x7f0c34027b30(Snoop/212995-00000007) [Aug 18 10:33:55] VERBOSE[13466] bridge_channel.c: Channel Snoop/212995-00000007 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:55] DEBUG[13465] app.c: play_and_record: , /var/spool/asterisk/recording/212999_qRuswmdCeCDHQrZUypTzhhPimeblcjmR, 'wav' [Aug 18 10:33:55] DEBUG[13465] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:55] VERBOSE[13465] app.c: x=0, open writing: /var/spool/asterisk/recording/212999_qRuswmdCeCDHQrZUypTzhhPimeblcjmR format: wav, 0x7f0c90040d20 [Aug 18 10:33:55] DEBUG[13466] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13466] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13466] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13463] channel.c: Channel 0x7f0c88086180 'Recorder/ARI-00000014;2' allocated [Aug 18 10:33:55] DEBUG[13463] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] DEBUG[13467] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13466] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13466] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13466] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5 is already using the new technology. [Aug 18 10:33:55] DEBUG[13466] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c34027b30(Snoop/212995-00000007) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13467] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13467] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13467] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13467] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13467] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13467] stasis.c: Creating topic. name: bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3, detail: [Aug 18 10:33:55] DEBUG[13461] stasis/control.c: robot_212995: Sending channel add_to_bridge command [Aug 18 10:33:55] DEBUG[13467] stasis.c: Topic 'bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3': 0x7f0c9c036820 created [Aug 18 10:33:55] DEBUG[13447] stasis/app.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[13468] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13467] stasis.c: Creating topic. name: cache:157/bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3, detail: [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13467] stasis.c: Topic 'cache:157/bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3': 0x7f0c9c03a6c0 created [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Finding handler for 0ec77f0c-7a86-4072-a1c4-e42f5256208c [Aug 18 10:33:55] DEBUG[13468] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pushing 0x7f0c8808e340(Recorder/ARI-00000014;2) [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13464] res_ari.c: No explicit handler found for 0ec77f0c-7a86-4072-a1c4-e42f5256208c. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13464] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13467] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13467] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13467] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13468] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:55] DEBUG[13467] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13467] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13467] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: calling simple_bridge technology constructor [Aug 18 10:33:55] VERBOSE[13468] bridge_channel.c: Channel Recorder/ARI-00000014;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:55] DEBUG[13467] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13468] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267'. Checking compatability for channels 'SIP/zvonobot-0000002e' and 'Recorder/ARI-00000014;2' [Aug 18 10:33:55] DEBUG[13468] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as could not get details [Aug 18 10:33:55] DEBUG[13468] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13468] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13468] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13468] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13468] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267 is already using the new technology. [Aug 18 10:33:55] DEBUG[13468] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13468] channel.c: Channel Recorder/ARI-00000014;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[13468] channel.c: Channel SIP/zvonobot-0000002e setting write format path: slin -> alaw [Aug 18 10:33:55] DEBUG[13468] channel.c: Channel SIP/zvonobot-0000002e setting read format path: alaw -> slin [Aug 18 10:33:55] DEBUG[13468] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13467] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP got report of 100 bytes from 178.62.121.41:10789 [Aug 18 10:33:55] DEBUG[13467] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13469] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13469] http.c: HTTP Request URI is /ari/channels/212999/snoop?app=calls_0&spy=in [Aug 18 10:33:55] DEBUG[13464] res_stasis_playback.c: 1629282833.94: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:55] DEBUG[13464] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13464] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13463] res_stasis_recording.c: 1629282835.130: Sending record(213012_IbmNZTtYydMrsTzVrzfNFHIyzKWXigCw.wav) command [Aug 18 10:33:55] DEBUG[13463] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] DEBUG[13470] app.c: play_and_record: , /var/spool/asterisk/recording/213012_IbmNZTtYydMrsTzVrzfNFHIyzKWXigCw, 'wav' [Aug 18 10:33:55] DEBUG[13470] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:55] VERBOSE[13470] app.c: x=0, open writing: /var/spool/asterisk/recording/213012_IbmNZTtYydMrsTzVrzfNFHIyzKWXigCw format: wav, 0x7f0ca4087820 [Aug 18 10:33:55] DEBUG[13463] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13472] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13469] http.c: match request [ari/channels/212999/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13472] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:55] DEBUG[13472] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13472] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13472] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13472] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13469] http.c: match request [ari/channels/212999/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13469] http.c: match request [ari/channels/212999/snoop] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13469] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13472] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13472] stasis.c: Creating topic. name: bridge:b7adaa29-9b73-48a7-8d8d-8ee58b870f71, detail: [Aug 18 10:33:55] DEBUG[13472] stasis.c: Topic 'bridge:b7adaa29-9b73-48a7-8d8d-8ee58b870f71': 0x7f0cb00987f0 created [Aug 18 10:33:55] DEBUG[13472] stasis.c: Creating topic. name: cache:158/bridge:b7adaa29-9b73-48a7-8d8d-8ee58b870f71, detail: [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Finding handler for channels/212999/snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13472] stasis.c: Topic 'cache:158/bridge:b7adaa29-9b73-48a7-8d8d-8ee58b870f71': 0x7f0cb00d3640 created [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Finding handler for 212999 [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channels create: Didn't match 212999 [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channels externalMedia: Didn't match 212999 [Aug 18 10:33:55] DEBUG[13469] res_ari.c: No explicit handler found for 212999. Using wildcard channelId. [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Finding handler for snoop [Aug 18 10:33:55] DEBUG[13472] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' can not use native RTP bridge as two channels are required [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:55] DEBUG[13472] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13472] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13472] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:55] DEBUG[13472] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13472] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13472] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13472] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13472] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13473] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13473] http.c: HTTP Request URI is /ari/channels/213012/snoop?app=calls_0&spy=in [Aug 18 10:33:55] DEBUG[13473] http.c: match request [ari/channels/213012/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13473] http.c: match request [ari/channels/213012/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13473] http.c: match request [ari/channels/213012/snoop] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13473] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Finding handler for channels/213012/snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Finding handler for 213012 [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channels create: Didn't match 213012 [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13469] stasis.c: Creating topic. name: channel:1629282835.132, detail: [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channels externalMedia: Didn't match 213012 [Aug 18 10:33:55] DEBUG[13473] res_ari.c: No explicit handler found for 213012. Using wildcard channelId. [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Finding handler for snoop [Aug 18 10:33:55] DEBUG[13469] stasis.c: Topic 'channel:1629282835.132': 0x7f0ca0058450 created [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] stasis.c: Creating topic. name: cache:159/channel:1629282835.132, detail: [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:55] DEBUG[13469] stasis.c: Topic 'cache:159/channel:1629282835.132': 0x7f0ca0058660 created [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:55] DEBUG[13473] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:55] DEBUG[13473] stasis.c: Creating topic. name: channel:1629282835.133, detail: [Aug 18 10:33:55] DEBUG[13469] channel.c: Channel 0x7f0ca0056680 'Snoop/212999-00000008' allocated [Aug 18 10:33:55] DEBUG[13473] stasis.c: Topic 'channel:1629282835.133': 0x7f0cac05cc80 created [Aug 18 10:33:55] DEBUG[13473] stasis.c: Creating topic. name: cache:160/channel:1629282835.133, detail: [Aug 18 10:33:55] DEBUG[13473] stasis.c: Topic 'cache:160/channel:1629282835.133': 0x7f0cac05d170 created [Aug 18 10:33:55] DEBUG[13454] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162'. Checking compatability for channels 'SIP/zvonobot-00000023' and 'Recorder/ARI-00000013;2' [Aug 18 10:33:55] DEBUG[13454] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as channel 'SIP/zvonobot-00000023' has features which prevent it [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13454] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13454] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 is already using the new technology. [Aug 18 10:33:55] DEBUG[13473] channel.c: Channel 0x7f0cac05bc90 'Snoop/213012-00000009' allocated [Aug 18 10:33:55] DEBUG[13459] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267'. Checking compatability for channels 'SIP/zvonobot-0000002e' and 'Recorder/ARI-00000014;2' [Aug 18 10:33:55] DEBUG[13459] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as channel 'SIP/zvonobot-0000002e' has features which prevent it [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13473] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13473] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13479] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13479] http.c: HTTP Request URI is /ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play?media=sound%3Asilence%2F2 [Aug 18 10:33:55] DEBUG[13479] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13479] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13479] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13479] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Finding handler for bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13459] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267 is already using the new technology. [Aug 18 10:33:55] DEBUG[13482] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13482] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213012&app=calls_0&format=slin16&external_host=127.0.0.1%3A50433 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13475] stasis/app.c: Channel '1629282835.133' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13482] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Finding handler for e2e70698-2279-429d-a48c-2fe9dd817267 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13457] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: No explicit handler found for e2e70698-2279-429d-a48c-2fe9dd817267. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:55] DEBUG[13457] stasis/control.c: robot_212995: Adding to bridge d177377e-a80b-4ad9-826a-cece7d5abce5 [Aug 18 10:33:55] DEBUG[13482] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13482] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13479] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13482] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13474] stasis/app.c: Channel '1629282835.132' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13457] stasis/app.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' is 3 interested in calls_0 [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:55] DEBUG[13474] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13469] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:55] DEBUG[13479] stasis.c: Creating topic. name: channel:1629282835.134, detail: [Aug 18 10:33:55] DEBUG[13469] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13483] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) is joining [Aug 18 10:33:55] DEBUG[13479] stasis.c: Topic 'channel:1629282835.134': 0x7f0c100699c0 created [Aug 18 10:33:55] DEBUG[13475] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13482] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:55] DEBUG[13482] netsock2.c: Splitting '127.0.0.1:50433' into... [Aug 18 10:33:55] DEBUG[13482] netsock2.c: ...host '127.0.0.1' and port '50433'. [Aug 18 10:33:55] DEBUG[13482] netsock2.c: Splitting '127.0.0.1:50433' into... [Aug 18 10:33:55] DEBUG[13482] netsock2.c: ...host '127.0.0.1' and port '50433'. [Aug 18 10:33:55] DEBUG[13482] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:55] DEBUG[13482] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c080871b0' [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) RTP allocated port 18144 [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) ICE creating session 127.0.0.1:18144 (18144) [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) ICE create [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) ICE add system candidates [Aug 18 10:33:55] DEBUG[13482] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:55] DEBUG[13482] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) ICE add candidate: 159.65.48.104:18144, 2130706431 [Aug 18 10:33:55] DEBUG[13482] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:55] DEBUG[13482] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:55] DEBUG[13482] res_rtp_asterisk.c: (0x7f0c080871b0) ICE add candidate: 10.131.0.10:18144, 2130706431 [Aug 18 10:33:55] DEBUG[13482] rtp_engine.c: RTP instance '0x7f0c080871b0' is setup and ready to go [Aug 18 10:33:55] DEBUG[13482] stasis.c: Creating topic. name: channel:robot_213012, detail: [Aug 18 10:33:55] DEBUG[13482] stasis.c: Topic 'channel:robot_213012': 0x7f0c08050ee0 created [Aug 18 10:33:55] DEBUG[13482] stasis.c: Creating topic. name: cache:162/channel:robot_213012, detail: [Aug 18 10:33:55] DEBUG[13482] stasis.c: Topic 'cache:162/channel:robot_213012': 0x7f0c08055370 created [Aug 18 10:33:55] DEBUG[13492] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13479] stasis.c: Creating topic. name: cache:161/channel:1629282835.134, detail: [Aug 18 10:33:55] DEBUG[13492] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212999&app=calls_0&format=slin16&external_host=127.0.0.1%3A50116 [Aug 18 10:33:55] DEBUG[13479] stasis.c: Topic 'cache:161/channel:1629282835.134': 0x7f0c10065140 created [Aug 18 10:33:55] DEBUG[13489] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13483] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: pushing 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) [Aug 18 10:33:55] DEBUG[13492] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:55] VERBOSE[13454] res_rtp_asterisk.c: 0x7f0c9800ef10 -- Strict RTP switching to RTP target address 178.62.121.41:16938 as source [Aug 18 10:33:55] DEBUG[13454] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:55] DEBUG[13454] channel.c: Channel SIP/zvonobot-00000023 setting read format path: ulaw -> slin [Aug 18 10:33:55] DEBUG[13489] http.c: HTTP Request URI is /ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play?media=sound%3Asilence%2F2 [Aug 18 10:33:55] DEBUG[13492] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13346] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13454] channel.c: Channel SIP/zvonobot-00000023 setting write format path: slin -> ulaw [Aug 18 10:33:55] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:33:55] DEBUG[13492] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13346] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13346] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13346] channel.c: Channel Announcer/ARI-0000000b;1 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13346] channel.c: Channel Announcer/ARI-0000000b;1 setting write format path: gsm -> slin [Aug 18 10:33:55] DEBUG[13346] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:55] VERBOSE[13346] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:55] VERBOSE[13483] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:55] DEBUG[13492] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 - start 1629282835.373298 answer 1629282835.393350 end 1629282835.680893 dur 0.307 bill 0.287 dispo ANSWERED [Aug 18 10:33:55] VERBOSE[13459] res_rtp_asterisk.c: 0x7f0c94023d30 -- Strict RTP switching to RTP target address 178.62.121.41:13636 as source [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:55] DEBUG[13459] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:55] DEBUG[13459] channel.c: Channel SIP/zvonobot-0000002e setting read format path: ulaw -> slin [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Finding handler for channels [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:55] DEBUG[13459] channel.c: Channel SIP/zvonobot-0000002e setting write format path: slin -> ulaw [Aug 18 10:33:55] DEBUG[13459] audiohook.c: Audiohook 0x7f0cac057320 has stale audio in its factories. Flushing them both [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:55] DEBUG[13489] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13489] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13489] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13489] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Finding handler for bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Finding handler for 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13383] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13383] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13383] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13383] channel.c: Channel Announcer/ARI-0000000e;1 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13383] channel.c: Channel 0x7f0c7803b230 'Announcer/ARI-0000000e;1' hanging up. Refs: 2 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:55] DEBUG[13489] res_ari.c: No explicit handler found for 3d1f9573-48d1-48a0-bcdd-9d7f09555162. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:55] DEBUG[13493] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13493] http.c: HTTP Request URI is /ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:55] DEBUG[13493] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13493] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13493] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13493] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Finding handler for bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e/play [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13494] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13494] http.c: HTTP Request URI is /ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Finding handler for 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13494] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [httpstatus] len 10 [Aug 18 10:33:55] DEBUG[13494] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [phoneprov] len 9 [Aug 18 10:33:55] DEBUG[13493] res_ari.c: No explicit handler found for 85c47f7b-0e24-4408-b7bf-5a532802bd8e. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13493] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13494] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play] with handler [ari] len 3 [Aug 18 10:33:55] DEBUG[13494] http.c: Match made with [ari] [Aug 18 10:33:55] DEBUG[13493] stasis.c: Creating topic. name: channel:1629282835.136, detail: [Aug 18 10:33:55] DEBUG[13483] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5'. Checking compatability for channels 'Snoop/212995-00000007' and 'UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40' [Aug 18 10:33:55] DEBUG[13483] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' can not use native RTP bridge as could not get details [Aug 18 10:33:55] DEBUG[13483] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13483] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13483] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13483] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13483] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5 is already using the new technology. [Aug 18 10:33:55] DEBUG[13483] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 setting read format path: slin16 -> slin16 [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Finding handler for bridges/d36cece3-ab54-488a-bcb0-0ed40691a344/play [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Finding handler for bridges [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:55] DEBUG[13489] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13493] stasis.c: Topic 'channel:1629282835.136': 0x7f0c2003b430 created [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:55] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13474] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel Snoop/212995-00000007 setting write format path: slin16 -> slin [Aug 18 10:33:55] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:55] DEBUG[13493] stasis.c: Creating topic. name: cache:163/channel:1629282835.136, detail: [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13475] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13492] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:55] DEBUG[13492] netsock2.c: Splitting '127.0.0.1:50116' into... [Aug 18 10:33:55] DEBUG[13493] stasis.c: Topic 'cache:163/channel:1629282835.136': 0x7f0c2003b580 created [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Finding handler for d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:33:55] DEBUG[13492] netsock2.c: ...host '127.0.0.1' and port '50116'. [Aug 18 10:33:55] DEBUG[13314] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel Snoop/212995-00000007 setting read format path: slin -> slin16 [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:55] DEBUG[13492] netsock2.c: Splitting '127.0.0.1:50116' into... [Aug 18 10:33:55] DEBUG[13494] res_ari.c: No explicit handler found for d36cece3-ab54-488a-bcb0-0ed40691a344. Using wildcard bridgeId. [Aug 18 10:33:55] DEBUG[13489] stasis.c: Creating topic. name: channel:1629282835.137, detail: [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 setting write format path: slin16 -> slin16 [Aug 18 10:33:55] DEBUG[13492] netsock2.c: ...host '127.0.0.1' and port '50116'. [Aug 18 10:33:55] DEBUG[13384] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13384] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13384] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:55] DEBUG[13384] channel.c: Channel Announcer/ARI-0000000f;1 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:55] DEBUG[13384] channel.c: Channel 0x7f0c80037a00 'Announcer/ARI-0000000f;1' hanging up. Refs: 2 [Aug 18 10:33:55] DEBUG[13492] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Finding handler for play [Aug 18 10:33:55] DEBUG[13492] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c24077280' [Aug 18 10:33:55] DEBUG[13489] stasis.c: Topic 'channel:1629282835.137': 0x7f0c18093d90 created [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) RTP allocated port 11806 [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:55] DEBUG[13494] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:55] DEBUG[13494] stasis.c: Creating topic. name: channel:1629282835.138, detail: [Aug 18 10:33:55] DEBUG[13494] stasis.c: Topic 'channel:1629282835.138': 0x7f0c2c06a150 created [Aug 18 10:33:55] DEBUG[13494] stasis.c: Creating topic. name: cache:165/channel:1629282835.138, detail: [Aug 18 10:33:55] DEBUG[13494] stasis.c: Topic 'cache:165/channel:1629282835.138': 0x7f0c2c018150 created [Aug 18 10:33:55] DEBUG[13489] stasis.c: Creating topic. name: cache:164/channel:1629282835.137, detail: [Aug 18 10:33:55] DEBUG[13479] channel.c: Channel 0x7f0c10068200 'Announcer/ARI-00000015;1' allocated [Aug 18 10:33:55] DEBUG[13383] channel.c: Channel 0x7f0c7803b230 'Announcer/ARI-0000000e;1' destroying [Aug 18 10:33:55] DEBUG[13374] bridge_channel.c: Setting 0x7f0c7803ac60(Announcer/ARI-0000000e;2) state from:0 to:1 [Aug 18 10:33:55] DEBUG[13489] stasis.c: Topic 'cache:164/channel:1629282835.137': 0x7f0c18090df0 created [Aug 18 10:33:55] DEBUG[13482] channel.c: Channel 0x7f0c08053190 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0' allocated [Aug 18 10:33:55] DEBUG[13489] channel.c: Channel 0x7f0c1808d1d0 'Announcer/ARI-00000017;1' allocated [Aug 18 10:33:55] DEBUG[13494] channel.c: Channel 0x7f0c2c070450 'Announcer/ARI-00000018;1' allocated [Aug 18 10:33:55] DEBUG[13493] channel.c: Channel 0x7f0c2005c6f0 'Announcer/ARI-00000016;1' allocated [Aug 18 10:33:55] DEBUG[13383] stasis.c: Destroying topic. name: cache:124/channel:1629282833.103, detail: [Aug 18 10:33:55] DEBUG[13383] stasis.c: Topic 'cache:124/channel:1629282833.103': 0x7f0c7803e120 destroyed [Aug 18 10:33:55] DEBUG[13383] stasis.c: Destroying topic. name: channel:1629282833.103, detail: [Aug 18 10:33:55] DEBUG[13383] stasis.c: Topic 'channel:1629282833.103': 0x7f0c7803d710 destroyed [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:55] DEBUG[13374] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pulling 0x7f0c7803ac60(Announcer/ARI-0000000e;2) [Aug 18 10:33:55] VERBOSE[13374] bridge_channel.c: Channel Announcer/ARI-0000000e;2 left 'softmix' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:33:55] DEBUG[13374] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7803ac60(Announcer/ARI-0000000e;2) is leaving softmix technology [Aug 18 10:33:55] DEBUG[13457] stasis/app.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' is 4 interested in calls_0 [Aug 18 10:33:55] DEBUG[13461] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:55] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12 instead [Aug 18 10:33:55] DEBUG[13461] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:33:55] DEBUG[13483] channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 setting write format path: slin -> slin16 [Aug 18 10:33:55] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP ooh, format changed from none to slin16 [Aug 18 10:33:55] DEBUG[13479] stasis.c: Creating topic. name: channel:1629282835.139, detail: [Aug 18 10:33:55] DEBUG[13479] stasis.c: Topic 'channel:1629282835.139': 0x7f0c1005db30 created [Aug 18 10:33:55] DEBUG[13479] stasis.c: Creating topic. name: cache:166/channel:1629282835.139, detail: [Aug 18 10:33:55] DEBUG[13384] channel.c: Channel 0x7f0c80037a00 'Announcer/ARI-0000000f;1' destroying [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:55] DEBUG[13482] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:55] VERBOSE[13482] res_rtp_asterisk.c: 0x7f0c0804b830 -- Strict RTP learning after remote address set to: 127.0.0.1:50433 [Aug 18 10:33:55] DEBUG[13479] stasis.c: Topic 'cache:166/channel:1629282835.139': 0x7f0c100712f0 created [Aug 18 10:33:55] DEBUG[13479] channel.c: Channel 0x7f0c1006f850 'Announcer/ARI-00000015;2' allocated [Aug 18 10:33:55] DEBUG[13479] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] DEBUG[13376] bridge_channel.c: Setting 0x7f0c80046b00(Announcer/ARI-0000000f;2) state from:0 to:1 [Aug 18 10:33:55] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:55] DEBUG[13479] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000015;1' [Aug 18 10:33:55] DEBUG[13496] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c1004ffb0(Announcer/ARI-00000015;2) is joining [Aug 18 10:33:55] DEBUG[13496] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pushing 0x7f0c1004ffb0(Announcer/ARI-00000015;2) [Aug 18 10:33:55] DEBUG[13374] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e'. Checking compatability for channels 'SIP/zvonobot-00000009' and 'Recorder/ARI-0000000d;2' [Aug 18 10:33:55] DEBUG[13482] res_stasis.c: calls_0: Subscribing to robot_213012 [Aug 18 10:33:55] DEBUG[13482] stasis/app.c: Channel 'robot_213012' is 1 interested in calls_0 [Aug 18 10:33:55] DEBUG[13374] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as channel 'SIP/zvonobot-00000009' has features which prevent it [Aug 18 10:33:55] DEBUG[13496] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] VERBOSE[13496] bridge_channel.c: Channel Announcer/ARI-00000015;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:55] DEBUG[13489] stasis.c: Creating topic. name: channel:1629282835.140, detail: [Aug 18 10:33:55] DEBUG[13489] stasis.c: Topic 'channel:1629282835.140': 0x7f0c18090c60 created [Aug 18 10:33:55] DEBUG[13489] stasis.c: Creating topic. name: cache:167/channel:1629282835.140, detail: [Aug 18 10:33:55] DEBUG[13489] stasis.c: Topic 'cache:167/channel:1629282835.140': 0x7f0c180bac50 created [Aug 18 10:33:55] DEBUG[13482] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:55] DEBUG[13489] channel.c: Channel 0x7f0c180b93f0 'Announcer/ARI-00000017;2' allocated [Aug 18 10:33:55] DEBUG[13489] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] DEBUG[13489] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000017;1' [Aug 18 10:33:55] DEBUG[13482] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13374] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13496] bridge.c: Chose bridge technology softmix [Aug 18 10:33:55] VERBOSE[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: switching from simple_bridge technology to softmix [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology constructor [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c9006b170(SIP/zvonobot-0000002e) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c8808e340(Recorder/ARI-00000014;2) to dummy bridge temporarily [Aug 18 10:33:55] VERBOSE[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: switching from softmix technology to simple_bridge [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology stop [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c1004ffb0(Announcer/ARI-00000015;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Announcer/ARI-00000015;2: [Aug 18 10:33:55] DEBUG[13496] channel.c: Channel Announcer/ARI-00000015;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] VERBOSE[13497] dial.c: Called 127.0.0.1:50433 [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13376] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pulling 0x7f0c80046b00(Announcer/ARI-0000000f;2) [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c7c01ea60(SIP/zvonobot-00000009) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13493] stasis.c: Creating topic. name: channel:1629282835.142, detail: [Aug 18 10:33:55] DEBUG[13493] stasis.c: Topic 'channel:1629282835.142': 0x7f0c2004d4e0 created [Aug 18 10:33:55] DEBUG[13493] stasis.c: Creating topic. name: cache:168/channel:1629282835.142, detail: [Aug 18 10:33:55] DEBUG[13493] stasis.c: Topic 'cache:168/channel:1629282835.142': 0x7f0c20040460 created [Aug 18 10:33:55] DEBUG[13493] channel.c: Channel 0x7f0c20061cb0 'Announcer/ARI-00000016;2' allocated [Aug 18 10:33:55] DEBUG[13493] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] DEBUG[13493] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000016;1' [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Announcer/ARI-00000015;2: [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Announcer/ARI-00000015;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is joining softmix technology [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: SIP/zvonobot-0000002e: [Aug 18 10:33:55] DEBUG[13475] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) ICE creating session 127.0.0.1:11806 (11806) [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) ICE create [Aug 18 10:33:55] DEBUG[13498] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c180a5170(Announcer/ARI-00000017;2) is joining [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c240520b0(Recorder/ARI-0000000d;2) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13384] stasis.c: Destroying topic. name: cache:125/channel:1629282833.104, detail: [Aug 18 10:33:55] DEBUG[13384] stasis.c: Topic 'cache:125/channel:1629282833.104': 0x7f0c8002e010 destroyed [Aug 18 10:33:55] DEBUG[13384] stasis.c: Destroying topic. name: channel:1629282833.104, detail: [Aug 18 10:33:55] DEBUG[13384] stasis.c: Topic 'channel:1629282833.104': 0x7f0c8002ddb0 destroyed [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: SIP/zvonobot-0000002e: [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: SIP/zvonobot-0000002e: Not in SFU mode [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Recorder/ARI-00000014;2: [Aug 18 10:33:55] DEBUG[13496] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13496] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Recorder/ARI-00000014;2: [Aug 18 10:33:55] DEBUG[13496] bridge_softmix.c: Recorder/ARI-00000014;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology start [Aug 18 10:33:55] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology destructor [Aug 18 10:33:55] VERBOSE[13376] bridge_channel.c: Channel Announcer/ARI-0000000f;2 left 'softmix' stasis-bridge [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is leaving softmix technology (dummy) [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is leaving softmix technology (dummy) [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology stop [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13376] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c80046b00(Announcer/ARI-0000000f;2) is leaving softmix technology [Aug 18 10:33:55] DEBUG[13494] stasis.c: Creating topic. name: channel:1629282835.141, detail: [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:55] DEBUG[13374] channel.c: Channel Recorder/ARI-0000000d;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[13376] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344'. Checking compatability for channels 'SIP/zvonobot-00000008' and 'Recorder/ARI-0000000c;2' [Aug 18 10:33:55] DEBUG[13376] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as channel 'SIP/zvonobot-00000008' has features which prevent it [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13376] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:55] VERBOSE[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: switching from softmix technology to simple_bridge [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology constructor [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c740224f0(SIP/zvonobot-00000008) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c1c00f210(Recorder/ARI-0000000c;2) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is leaving softmix technology (dummy) [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is leaving softmix technology (dummy) [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology stop [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13374] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13376] channel.c: Channel Recorder/ARI-0000000c;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) ICE add system candidates [Aug 18 10:33:55] VERBOSE[13497] dial.c: UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 answered [Aug 18 10:33:55] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50433 [Aug 18 10:33:55] DEBUG[13498] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pushing 0x7f0c180a5170(Announcer/ARI-00000017;2) [Aug 18 10:33:55] DEBUG[13479] res_stasis_playback.c: 1629282835.134: Sending play(sound:silence/2) command [Aug 18 10:33:55] DEBUG[13479] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:55] VERBOSE[13497] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 [Aug 18 10:33:55] DEBUG[13479] http.c: HTTP closing session. Top level [Aug 18 10:33:55] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:55] DEBUG[13492] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:55] DEBUG[13492] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) ICE add candidate: 159.65.48.104:11806, 2130706431 [Aug 18 10:33:55] DEBUG[13492] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:55] DEBUG[13492] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:55] DEBUG[13492] res_rtp_asterisk.c: (0x7f0c24077280) ICE add candidate: 10.131.0.10:11806, 2130706431 [Aug 18 10:33:55] DEBUG[13492] rtp_engine.c: RTP instance '0x7f0c24077280' is setup and ready to go [Aug 18 10:33:55] DEBUG[13492] stasis.c: Creating topic. name: channel:robot_212999, detail: [Aug 18 10:33:55] DEBUG[13497] stasis/app.c: Channel 'robot_213012' is 2 interested in calls_0 [Aug 18 10:33:55] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50433 - state 2 (In use) [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50433, detail: [Aug 18 10:33:55] DEBUG[13494] stasis.c: Topic 'channel:1629282835.141': 0x7f0c2c060790 created [Aug 18 10:33:55] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50433': 0x7f0c840684e0 created [Aug 18 10:33:55] DEBUG[13499] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: starting mixing thread [Aug 18 10:33:55] DEBUG[13492] stasis.c: Topic 'channel:robot_212999': 0x7f0c240f2580 created [Aug 18 10:33:55] DEBUG[13374] channel.c: Channel Recorder/ARI-0000000d;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50433' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:55] DEBUG[13376] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13492] stasis.c: Creating topic. name: cache:170/channel:robot_212999, detail: [Aug 18 10:33:55] DEBUG[13501] channel.c: Channel Announcer/ARI-00000015;1 setting write format path: gsm -> slin [Aug 18 10:33:55] DEBUG[13494] stasis.c: Creating topic. name: cache:169/channel:1629282835.141, detail: [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining simple_bridge technology [Aug 18 10:33:55] DEBUG[13498] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:55] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:55] DEBUG[13376] channel.c: Channel Recorder/ARI-0000000c;2 setting read format path: slin -> slin [Aug 18 10:33:55] DEBUG[13374] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13492] stasis.c: Topic 'cache:170/channel:robot_212999': 0x7f0c240f27e0 created [Aug 18 10:33:55] DEBUG[13501] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:55] VERBOSE[13498] bridge_channel.c: Channel Announcer/ARI-00000017;2 joined 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: deferring softmix technology destructor [Aug 18 10:33:55] DEBUG[13374] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: queueing action type:13 sub:1000 [Aug 18 10:33:55] DEBUG[13376] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:33:55] VERBOSE[13501] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology start [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: deferring softmix technology destructor [Aug 18 10:33:55] DEBUG[13376] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: queueing action type:13 sub:1000 [Aug 18 10:33:55] DEBUG[13494] stasis.c: Topic 'cache:169/channel:1629282835.141': 0x7f0c2c065ed0 created [Aug 18 10:33:55] DEBUG[13459] audiohook.c: Audiohook 0x7f0cac057320 has stale audio in its factories. Flushing them both [Aug 18 10:33:55] DEBUG[13353] channel.c: Recorder/ARI-0000000d;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:55] DEBUG[13380] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: stopping mixing thread [Aug 18 10:33:55] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:55] DEBUG[20534] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: Waiting for mixing thread to die. [Aug 18 10:33:55] DEBUG[13349] res_rtp_asterisk.c: (0x7f0c70012180) RTP audio difference is 848, ms is 126 [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP ooh, format changed from none to ulaw [Aug 18 10:33:55] DEBUG[13349] channel.c: SIP/zvonobot-00000009: Dropping redundant connected line update "" <>. [Aug 18 10:33:55] DEBUG[13352] channel.c: Recorder/ARI-0000000c;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:55] DEBUG[13374] channel.c: Channel 0x7f0c78042670 'Announcer/ARI-0000000e;2' hanging up. Refs: 2 [Aug 18 10:33:55] DEBUG[13503] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c200534f0(Announcer/ARI-00000016;2) is joining [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:55] DEBUG[13503] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pushing 0x7f0c200534f0(Announcer/ARI-00000016;2) [Aug 18 10:33:55] DEBUG[13498] bridge.c: Chose bridge technology softmix [Aug 18 10:33:55] VERBOSE[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: switching from simple_bridge technology to softmix [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology constructor [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0ca0053060(SIP/zvonobot-00000023) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0c7c018d60(Recorder/ARI-00000013;2) to dummy bridge temporarily [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology stop [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c180a5170(Announcer/ARI-00000017;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: Announcer/ARI-00000017;2: [Aug 18 10:33:55] DEBUG[13498] channel.c: Channel Announcer/ARI-00000017;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13347] channel.c: SIP/zvonobot-00000008: Dropping redundant connected line update "" <>. [Aug 18 10:33:55] DEBUG[13376] channel.c: Channel 0x7f0c8003faf0 'Announcer/ARI-0000000f;2' hanging up. Refs: 2 [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: Announcer/ARI-00000017;2: [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: Announcer/ARI-00000017;2: Not in SFU mode [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining softmix technology [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: SIP/zvonobot-00000023: [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: SIP/zvonobot-00000023: [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: SIP/zvonobot-00000023: Not in SFU mode [Aug 18 10:33:55] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining softmix technology [Aug 18 10:33:55] DEBUG[13498] bridge_softmix.c: Recorder/ARI-00000013;2: [Aug 18 10:33:55] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:55] DEBUG[20534] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: Waiting for mixing thread to die. [Aug 18 10:33:55] DEBUG[13378] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: stopping mixing thread [Aug 18 10:33:55] DEBUG[13498] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:33:55] DEBUG[13502] http.c: HTTP opening session. Top level [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:55] DEBUG[13498] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:55] DEBUG[13503] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:55] VERBOSE[13503] bridge_channel.c: Channel Announcer/ARI-00000016;2 joined 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:33:56] DEBUG[13498] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13498] bridge_softmix.c: Recorder/ARI-00000013;2: [Aug 18 10:33:55] DEBUG[13502] http.c: HTTP Request URI is /ari/bridges/b7adaa29-9b73-48a7-8d8d-8ee58b870f71/addChannel?channel=1629282835.133%2Crobot_213012 [Aug 18 10:33:56] DEBUG[13498] bridge_softmix.c: Recorder/ARI-00000013;2: Not in SFU mode [Aug 18 10:33:56] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology start [Aug 18 10:33:56] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology destructor [Aug 18 10:33:56] DEBUG[13374] channel.c: Channel 0x7f0c78042670 'Announcer/ARI-0000000e;2' destroying [Aug 18 10:33:56] DEBUG[13376] channel.c: Channel 0x7f0c8003faf0 'Announcer/ARI-0000000f;2' destroying [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13374] stasis.c: Destroying topic. name: cache:126/channel:1629282833.105, detail: [Aug 18 10:33:56] DEBUG[13374] stasis.c: Topic 'cache:126/channel:1629282833.105': 0x7f0c7800a070 destroyed [Aug 18 10:33:56] DEBUG[13494] channel.c: Channel 0x7f0c2c075a10 'Announcer/ARI-00000018;2' allocated [Aug 18 10:33:56] DEBUG[13494] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:56] DEBUG[13374] stasis.c: Destroying topic. name: channel:1629282833.105, detail: [Aug 18 10:33:56] DEBUG[13494] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000018;1' [Aug 18 10:33:56] DEBUG[13492] channel.c: Channel 0x7f0c240f0910 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280' allocated [Aug 18 10:33:56] DEBUG[13492] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:56] VERBOSE[13492] res_rtp_asterisk.c: 0x7f0c24086d40 -- Strict RTP learning after remote address set to: 127.0.0.1:50116 [Aug 18 10:33:56] DEBUG[13505] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c2c0538b0(Announcer/ARI-00000018;2) is joining [Aug 18 10:33:56] DEBUG[13374] stasis.c: Topic 'channel:1629282833.105': 0x7f0c780398b0 destroyed [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13505] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pushing 0x7f0c2c0538b0(Announcer/ARI-00000018;2) [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13503] bridge.c: Chose bridge technology softmix [Aug 18 10:33:56] VERBOSE[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: switching from simple_bridge technology to softmix [Aug 18 10:33:56] DEBUG[13505] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:56] VERBOSE[13505] bridge_channel.c: Channel Announcer/ARI-00000018;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology constructor [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c7c01ea60(SIP/zvonobot-00000009) to dummy bridge temporarily [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c240520b0(Recorder/ARI-0000000d;2) to dummy bridge temporarily [Aug 18 10:33:56] DEBUG[13492] res_stasis.c: calls_0: Subscribing to robot_212999 [Aug 18 10:33:56] DEBUG[13502] http.c: match request [ari/bridges/b7adaa29-9b73-48a7-8d8d-8ee58b870f71/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is leaving simple_bridge technology (dummy) [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13492] stasis/app.c: Channel 'robot_212999' is 1 interested in calls_0 [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology stop [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c200534f0(Announcer/ARI-00000016;2) is joining softmix technology [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Announcer/ARI-00000016;2: [Aug 18 10:33:56] DEBUG[13503] channel.c: Channel Announcer/ARI-00000016;2 setting write format path: slin -> slin [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Announcer/ARI-00000016;2: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Announcer/ARI-00000016;2: Not in SFU mode [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is joining softmix technology [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: SIP/zvonobot-00000009: [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: SIP/zvonobot-00000009: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: SIP/zvonobot-00000009: Not in SFU mode [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining softmix technology [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Recorder/ARI-0000000d;2: [Aug 18 10:33:56] DEBUG[13503] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13503] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Recorder/ARI-0000000d;2: [Aug 18 10:33:56] DEBUG[13503] bridge_softmix.c: Recorder/ARI-0000000d;2: Not in SFU mode [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology start [Aug 18 10:33:56] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology destructor [Aug 18 10:33:56] DEBUG[13492] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:56] DEBUG[13492] http.c: HTTP closing session. Top level [Aug 18 10:33:56] DEBUG[13505] bridge.c: Chose bridge technology softmix [Aug 18 10:33:56] VERBOSE[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: switching from simple_bridge technology to softmix [Aug 18 10:33:56] DEBUG[13507] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: starting mixing thread [Aug 18 10:33:56] DEBUG[13493] res_stasis_playback.c: 1629282835.136: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:56] DEBUG[13493] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:56] DEBUG[13493] http.c: HTTP closing session. Top level [Aug 18 10:33:56] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology constructor [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c740224f0(SIP/zvonobot-00000008) to dummy bridge temporarily [Aug 18 10:33:56] DEBUG[13513] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c1c00f210(Recorder/ARI-0000000c;2) to dummy bridge temporarily [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is leaving simple_bridge technology (dummy) [Aug 18 10:33:56] DEBUG[13376] stasis.c: Destroying topic. name: cache:128/channel:1629282833.107, detail: [Aug 18 10:33:56] DEBUG[13502] http.c: match request [ari/bridges/b7adaa29-9b73-48a7-8d8d-8ee58b870f71/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13376] stasis.c: Topic 'cache:128/channel:1629282833.107': 0x7f0c80039ae0 destroyed [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:56] DEBUG[13502] http.c: match request [ari/bridges/b7adaa29-9b73-48a7-8d8d-8ee58b870f71/addChannel] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13502] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Finding handler for bridges/b7adaa29-9b73-48a7-8d8d-8ee58b870f71/addChannel [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Finding handler for bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Finding handler for b7adaa29-9b73-48a7-8d8d-8ee58b870f71 [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13502] res_ari.c: No explicit handler found for b7adaa29-9b73-48a7-8d8d-8ee58b870f71. Using wildcard bridgeId. [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Finding handler for addChannel [Aug 18 10:33:56] DEBUG[13502] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:56] DEBUG[13502] stasis/control.c: 1629282835.133: Sending channel add_to_bridge command [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology stop [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c2c0538b0(Announcer/ARI-00000018;2) is joining softmix technology [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:56] DEBUG[13376] stasis.c: Destroying topic. name: channel:1629282833.107, detail: [Aug 18 10:33:56] DEBUG[13376] stasis.c: Topic 'channel:1629282833.107': 0x7f0c80041960 destroyed [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:56] VERBOSE[13508] dial.c: Called 127.0.0.1:50116 [Aug 18 10:33:56] DEBUG[13499] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:56] DEBUG[13509] channel.c: Channel Announcer/ARI-00000016;1 setting write format path: gsm -> slin [Aug 18 10:33:56] DEBUG[13509] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:56] VERBOSE[13509] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Announcer/ARI-00000018;2: [Aug 18 10:33:56] DEBUG[13475] bridge_roles.c: Roles did not exist on channel Snoop/213012-00000009 [Aug 18 10:33:56] DEBUG[13475] stasis/control.c: 1629282835.133: Adding to bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 [Aug 18 10:33:56] DEBUG[13475] stasis/app.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' is 1 interested in calls_0 [Aug 18 10:33:56] DEBUG[13513] http.c: HTTP Request URI is /ari/channels/213034?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117006&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13513] http.c: match request [ari/channels/213034] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13513] http.c: match request [ari/channels/213034] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13513] http.c: match request [ari/channels/213034] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13517] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13518] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x2c12c90(Snoop/213012-00000009) is joining [Aug 18 10:33:56] DEBUG[13513] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13517] http.c: HTTP Request URI is /ari/channels/213035?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117005&callerId=74950493843 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:33:56] DEBUG[13517] http.c: match request [ari/channels/213035] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13517] http.c: match request [ari/channels/213035] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13517] http.c: match request [ari/channels/213035] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13517] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13513] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13517] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Finding handler for channels/213035 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13505] channel.c: Channel Announcer/ARI-00000018;2 setting write format path: slin -> slin [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Announcer/ARI-00000018;2: [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Finding handler for channels/213034 [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Announcer/ARI-00000018;2: Not in SFU mode [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] http.c: HTTP Request URI is /ari/channels/213036?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117004&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6979eb32;received=159.65.48.104 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 776884208 776884208 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14674 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6979eb32;received=159.65.48.104 [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Finding handler for 213034 [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking channels create: Didn't match 213034 [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13513] res_ari.c: Checking channels externalMedia: Didn't match 213034 [Aug 18 10:33:56] DEBUG[13513] res_ari.c: No explicit handler found for 213034. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34a9f263 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is joining softmix technology [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:56] DEBUG[13522] http.c: match request [ari/channels/213036] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13522] http.c: match request [ari/channels/213036] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: SIP/zvonobot-00000008: [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: SIP/zvonobot-00000008: [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: SIP/zvonobot-00000008: Not in SFU mode [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining softmix technology [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Recorder/ARI-0000000c;2: [Aug 18 10:33:56] DEBUG[13505] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:56] DEBUG[13505] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Recorder/ARI-0000000c;2: [Aug 18 10:33:56] DEBUG[13505] bridge_softmix.c: Recorder/ARI-0000000c;2: Not in SFU mode [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology start [Aug 18 10:33:56] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology destructor [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 776884208 776884208 IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14674 RTP/AVP 0 8 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag as34a9f263 [Aug 18 10:33:56] DEBUG[13518] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: pushing 0x2c12c90(Snoop/213012-00000009) [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50116 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Finding handler for 213035 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking channels create: Didn't match 213035 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Stopping retransmission on '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:56] DEBUG[13522] http.c: match request [ari/channels/213036] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13497] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 [Aug 18 10:33:56] DEBUG[13522] http.c: Match made with [ari] [Aug 18 10:33:56] VERBOSE[13508] dial.c: UnicastRTP/127.0.0.1:50116-0x7f0c24077280 answered [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:56] DEBUG[13499] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[13528] http.c: HTTP opening session. Top level [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Got SDP version 776884208 and unique parts [root 776884208 IN IP4 178.62.121.41] [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 776884208 776884208 IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE set role failed; no ice instance [Aug 18 10:33:56] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP setting address on RTP instance [Aug 18 10:33:56] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c7c0228e0 -- Strict RTP learning after remote address set to: 178.62.121.41:14674 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14674 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00c6bc8) from 0x7f0c147e2330 to 0x7f0c7c020f68 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00b1688) from 0x7f0c147e2330 to 0x7f0c7c020f68 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0062498) from 0x7f0c147e2330 to 0x7f0c7c020f68 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP ignoring duplicate property [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002b setting read format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002b setting write format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS - ast_rtp_activate rtp=0x7f0c7c0228e0 - setup and perform DTLS' [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0228e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0228e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:56] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Strict routing enforced for session 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8f8bf9 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Contact: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session timer started: 19 - 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 1768000ms [Aug 18 10:33:56] DEBUG[13504] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: starting mixing thread [Aug 18 10:33:56] VERBOSE[13518] bridge_channel.c: Channel Snoop/213012-00000009 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:56] VERBOSE[13183] dial.c: SIP/zvonobot-0000002b answered [Aug 18 10:33:56] DEBUG[13489] res_stasis_playback.c: 1629282835.137: Sending play(sound:silence/2) command [Aug 18 10:33:56] DEBUG[13528] http.c: HTTP Request URI is /ari/channels/213037?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117003&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13528] http.c: match request [ari/channels/213037] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13528] http.c: match request [ari/channels/213037] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50116 - state 2 (In use) [Aug 18 10:33:56] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50116, detail: [Aug 18 10:33:56] DEBUG[13489] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:56] DEBUG[13517] res_ari.c: Checking channels externalMedia: Didn't match 213035 [Aug 18 10:33:56] DEBUG[13528] http.c: match request [ari/channels/213037] with handler [ari] len 3 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02e5764a;received=159.65.48.104 From: ;tag=as0b424b33 To: ;tag=as1f220605 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1812771302 1812771302 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15418 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02e5764a;received=159.65.48.104 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0b424b33 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1f220605 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1812771302 1812771302 IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15418 RTP/AVP 0 8 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 (Checking To) --From tag as0b424b33 --To-tag as1f220605 [Aug 18 10:33:56] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50116': 0x7f0c84080730 created [Aug 18 10:33:56] DEBUG[13522] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13528] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13489] http.c: HTTP closing session. Top level [Aug 18 10:33:56] VERBOSE[13183] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000002b [Aug 18 10:33:56] DEBUG[13517] res_ari.c: No explicit handler found for 213035. Using wildcard channelId. [Aug 18 10:33:56] VERBOSE[13508] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50116-0x7f0c24077280 [Aug 18 10:33:56] DEBUG[13528] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Finding handler for channels/213037 [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Finding handler for 213037 [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking channels create: Didn't match 213037 [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13528] res_ari.c: Checking channels externalMedia: Didn't match 213037 [Aug 18 10:33:56] DEBUG[13528] res_ari.c: No explicit handler found for 213037. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:56] DEBUG[13531] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Stopping retransmission on '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Got SDP version 1812771302 and unique parts [root 1812771302 IN IP4 178.62.121.41] [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1812771302 1812771302 IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:56] DEBUG[13534] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13534] http.c: HTTP Request URI is /ari/channels/213041?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116999&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Finding handler for channels/213036 [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Finding handler for 213036 [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking channels create: Didn't match 213036 [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13522] res_ari.c: Checking channels externalMedia: Didn't match 213036 [Aug 18 10:33:56] DEBUG[13522] res_ari.c: No explicit handler found for 213036. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[13183] stasis/app.c: Channel '213008' is 2 interested in calls_0 [Aug 18 10:33:56] DEBUG[13534] http.c: match request [ari/channels/213041] with handler [httpstatus] len 10 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13534] http.c: match request [ari/channels/213041] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:56] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:33:56] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50116' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: Allocating new SIP dialog for 3fa9e78873a6cc580a601819030c0c9c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13528] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c04bc00' [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) RTP allocated port 18980 [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE creating session 0.0.0.0:18980 (18980) [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE create [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE add system candidates [Aug 18 10:33:56] DEBUG[13528] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13528] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE add candidate: 159.65.48.104:18980, 2130706431 [Aug 18 10:33:56] DEBUG[13528] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13528] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE add candidate: 10.131.0.10:18980, 2130706431 [Aug 18 10:33:56] DEBUG[13528] rtp_engine.c: RTP instance '0x7f0c9c04bc00' is setup and ready to go [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE stopped [Aug 18 10:33:56] DEBUG[13528] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13528] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13528] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13528] res_rtp_asterisk.c: (0x7f0c9c04bc00) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13528] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13528] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:56] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:56] DEBUG[13531] http.c: HTTP Request URI is /ari/channels/213038?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117002&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:56] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP ooh, format changed from none to ulaw [Aug 18 10:33:56] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13532] channel.c: Channel Announcer/ARI-00000017;1 setting write format path: gsm -> slin [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13535] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13535] http.c: HTTP Request URI is /ari/channels/213039?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117001&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: SIP call-id changed from '3fa9e78873a6cc580a601819030c0c9c@127.0.1.1:5060' to '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13528] stasis.c: Creating topic. name: channel:213037, detail: [Aug 18 10:33:56] DEBUG[13528] stasis.c: Topic 'channel:213037': 0x7f0c9c03dfb0 created [Aug 18 10:33:56] DEBUG[13528] stasis.c: Creating topic. name: cache:171/channel:213037, detail: [Aug 18 10:33:56] DEBUG[13528] stasis.c: Topic 'cache:171/channel:213037': 0x7f0c9c044240 created [Aug 18 10:33:56] DEBUG[13534] http.c: match request [ari/channels/213041] with handler [ari] len 3 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:56] DEBUG[13494] res_stasis_playback.c: 1629282835.138: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:56] DEBUG[13512] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: starting mixing thread [Aug 18 10:33:56] DEBUG[13531] http.c: match request [ari/channels/213038] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13494] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:56] DEBUG[13494] http.c: HTTP closing session. Top level [Aug 18 10:33:56] DEBUG[13535] http.c: match request [ari/channels/213039] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:56] DEBUG[13538] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE set role failed; no ice instance [Aug 18 10:33:56] DEBUG[13531] http.c: match request [ari/channels/213038] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13531] http.c: match request [ari/channels/213038] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP setting address on RTP instance [Aug 18 10:33:56] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c9c00a550 -- Strict RTP learning after remote address set to: 178.62.121.41:15418 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:15418 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00cc8e8) from 0x7f0c147e2330 to 0x7f0c9c008c08 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00cc998) from 0x7f0c147e2330 to 0x7f0c9c008c08 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00d11c8) from 0x7f0c147e2330 to 0x7f0c9c008c08 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP ignoring duplicate property [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:56] DEBUG[13531] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13531] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13536] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:56] DEBUG[13536] http.c: HTTP Request URI is /ari/channels/213040?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821117000&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13536] http.c: match request [ari/channels/213040] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13536] http.c: match request [ari/channels/213040] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13535] http.c: match request [ari/channels/213039] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13535] http.c: match request [ari/channels/213039] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13538] http.c: HTTP Request URI is /ari/channels/213043?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116997&callerId=74950493843 [Aug 18 10:33:56] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13538] http.c: match request [ari/channels/213043] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000e setting read format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[13535] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13535] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Finding handler for channels/213039 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Finding handler for channels/213038 [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13539] http.c: HTTP opening session. Top level [Aug 18 10:33:56] DEBUG[13314] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000e setting write format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS - ast_rtp_activate rtp=0x7f0c9c00a550 - setup and perform DTLS' [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c00a550) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c00a550) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:56] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117063@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07d9a7d4 Max-Forwards: 70 From: ;tag=as0b424b33 To: ;tag=as1f220605 Contact: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session timer started: 25 - 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 1768000ms [Aug 18 10:33:56] DEBUG[13518] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' can not use native RTP bridge as two channels are required [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:56] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13534] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13518] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:56] VERBOSE[12956] dial.c: SIP/zvonobot-0000000e answered [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[12956] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000000e [Aug 18 10:33:56] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13518] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:56] DEBUG[13538] http.c: match request [ari/channels/213043] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK417bdd1e;received=159.65.48.104 From: ;tag=as08bf07d1 To: ;tag=as26319206 Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 812455523 812455523 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18792 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 3048, ms is 401 [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13538] http.c: match request [ari/channels/213043] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Finding handler for 213038 [Aug 18 10:33:56] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13532] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:56] VERBOSE[13532] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: Allocating new SIP dialog for 3482fba6495368d56526d27e6e5f3aac@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13522] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca804b700' [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) RTP allocated port 18840 [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE creating session 0.0.0.0:18840 (18840) [Aug 18 10:33:56] DEBUG[13539] http.c: HTTP Request URI is /ari/channels/213042?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116998&callerId=74950493843 [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:56] DEBUG[13518] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13518] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking channels create: Didn't match 213038 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:56] DEBUG[13537] channel.c: Channel Announcer/ARI-00000018;1 setting write format path: gsm -> slin [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13531] res_ari.c: Checking channels externalMedia: Didn't match 213038 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:56] DEBUG[13534] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13536] http.c: match request [ari/channels/213040] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13536] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTCP got report of 100 bytes from 178.62.121.41:18327 [Aug 18 10:33:56] DEBUG[13536] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13518] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 is already using the new technology. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK417bdd1e;received=159.65.48.104 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08bf07d1 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as26319206 [Aug 18 10:33:56] DEBUG[13531] res_ari.c: No explicit handler found for 213038. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Finding handler for channels/213041 [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE create [Aug 18 10:33:56] DEBUG[13538] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] http.c: match request [ari/channels/213042] with handler [httpstatus] len 10 [Aug 18 10:33:56] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Finding handler for 213041 [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking channels create: Didn't match 213041 [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13534] res_ari.c: Checking channels externalMedia: Didn't match 213041 [Aug 18 10:33:56] DEBUG[13534] res_ari.c: No explicit handler found for 213041. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13537] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:56] DEBUG[13518] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x2c12c90(Snoop/213012-00000009) is joining simple_bridge technology [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE add system candidates [Aug 18 10:33:56] DEBUG[13522] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13522] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE add candidate: 159.65.48.104:18840, 2130706431 [Aug 18 10:33:56] DEBUG[13522] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13522] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE add candidate: 10.131.0.10:18840, 2130706431 [Aug 18 10:33:56] DEBUG[13522] rtp_engine.c: RTP instance '0x7f0ca804b700' is setup and ready to go [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) ICE stopped [Aug 18 10:33:56] DEBUG[13522] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13522] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13522] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13522] res_rtp_asterisk.c: (0x7f0ca804b700) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13522] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13522] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: SIP call-id changed from '3482fba6495368d56526d27e6e5f3aac@127.0.1.1:5060' to '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13522] stasis.c: Creating topic. name: channel:213036, detail: [Aug 18 10:33:56] DEBUG[13522] stasis.c: Topic 'channel:213036': 0x7f0ca805dad0 created [Aug 18 10:33:56] DEBUG[13522] stasis.c: Creating topic. name: cache:172/channel:213036, detail: [Aug 18 10:33:56] DEBUG[13522] stasis.c: Topic 'cache:172/channel:213036': 0x7f0ca805e500 created [Aug 18 10:33:56] DEBUG[13538] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] http.c: match request [ari/channels/213042] with handler [phoneprov] len 9 [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[12956] stasis/app.c: Channel '212977' is 2 interested in calls_0 [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] http.c: match request [ari/channels/213042] with handler [ari] len 3 [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: Allocating new SIP dialog for 3c0bcba9240f04b4420fbcab37571bdf@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13517] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c94050050' [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) RTP allocated port 10238 [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE creating session 0.0.0.0:10238 (10238) [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE create [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE add system candidates [Aug 18 10:33:56] DEBUG[13517] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13517] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE add candidate: 159.65.48.104:10238, 2130706431 [Aug 18 10:33:56] DEBUG[13517] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13517] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE add candidate: 10.131.0.10:10238, 2130706431 [Aug 18 10:33:56] DEBUG[13517] rtp_engine.c: RTP instance '0x7f0c94050050' is setup and ready to go [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) ICE stopped [Aug 18 10:33:56] DEBUG[13517] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13517] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13517] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13517] res_rtp_asterisk.c: (0x7f0c94050050) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13517] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13517] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13517] chan_sip.c: SIP call-id changed from '3c0bcba9240f04b4420fbcab37571bdf@127.0.1.1:5060' to '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13517] stasis.c: Creating topic. name: channel:213035, detail: [Aug 18 10:33:56] DEBUG[13517] stasis.c: Topic 'channel:213035': 0x7f0c9405eed0 created [Aug 18 10:33:56] DEBUG[13517] stasis.c: Creating topic. name: cache:173/channel:213035, detail: [Aug 18 10:33:56] DEBUG[13517] stasis.c: Topic 'cache:173/channel:213035': 0x7f0c9405f900 created [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Finding handler for channels/213040 [Aug 18 10:33:56] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:56] DEBUG[13508] stasis/app.c: Channel 'robot_212999' is 2 interested in calls_0 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13539] http.c: Match made with [ari] [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Finding handler for channels/213043 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Finding handler for 213039 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking channels create: Didn't match 213039 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13535] res_ari.c: Checking channels externalMedia: Didn't match 213039 [Aug 18 10:33:56] DEBUG[13535] res_ari.c: No explicit handler found for 213039. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13502] stasis/control.c: robot_213012: Sending channel add_to_bridge command [Aug 18 10:33:56] DEBUG[13475] stasis/app.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' is 2 interested in calls_0 [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Finding handler for 213040 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 812455523 812455523 IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18792 RTP/AVP 0 8 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:56] VERBOSE[13537] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:56] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 18 instead [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking channels create: Didn't match 213040 [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13536] res_ari.c: Checking channels externalMedia: Didn't match 213040 [Aug 18 10:33:56] DEBUG[13536] res_ari.c: No explicit handler found for 213040. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[13539] http.c: HTTP consuming request body [Aug 18 10:33:56] DEBUG[13528] channel.c: Channel 0x7f0c9c03d0a0 'SIP/zvonobot-00000046' allocated [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:56] DEBUG[13528] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13349] res_rtp_asterisk.c: (0x7f0c70012180) RTP audio difference is 776, ms is 117 [Aug 18 10:33:56] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: = Looking for Call ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 (Checking To) --From tag as08bf07d1 --To-tag as26319206 [Aug 18 10:33:56] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: Allocating new SIP dialog for 111105b7058ce6215acff8936326c58f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13535] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb00e8390' [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) RTP allocated port 10382 [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: Allocating new SIP dialog for 4fb6dbfd071566f359ddd96c44ad7f78@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13531] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c980a1440' [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) RTP allocated port 17334 [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE creating session 0.0.0.0:17334 (17334) [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE create [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE add system candidates [Aug 18 10:33:56] DEBUG[13531] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Finding handler for channels/213042 [Aug 18 10:33:56] DEBUG[13314] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Finding handler for channels [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:56] DEBUG[13531] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13507] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13528] res_stasis.c: calls_0: Subscribing to 213037 [Aug 18 10:33:56] DEBUG[13528] stasis/app.c: Channel '213037' is 1 interested in calls_0 [Aug 18 10:33:56] DEBUG[13528] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:56] DEBUG[13528] http.c: HTTP closing session. Top level [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Outgoing Call for 79821117003 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE add candidate: 159.65.48.104:17334, 2130706431 [Aug 18 10:33:56] DEBUG[13531] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13531] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE add candidate: 10.131.0.10:17334, 2130706431 [Aug 18 10:33:56] DEBUG[13531] rtp_engine.c: RTP instance '0x7f0c980a1440' is setup and ready to go [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) ICE stopped [Aug 18 10:33:56] DEBUG[13531] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13531] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13531] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13531] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13531] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13531] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13531] chan_sip.c: SIP call-id changed from '4fb6dbfd071566f359ddd96c44ad7f78@127.0.1.1:5060' to '1591cc604083d1a612552226202481e2@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Finding handler for 213042 [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking channels create: Didn't match 213042 [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13539] res_ari.c: Checking channels externalMedia: Didn't match 213042 [Aug 18 10:33:56] DEBUG[13539] res_ari.c: No explicit handler found for 213042. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Finding handler for 213043 [Aug 18 10:33:56] DEBUG[13522] channel.c: Channel 0x7f0ca805b960 'SIP/zvonobot-00000047' allocated [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:56] DEBUG[13522] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE creating session 0.0.0.0:10382 (10382) [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13531] stasis.c: Creating topic. name: channel:213038, detail: [Aug 18 10:33:56] DEBUG[13531] stasis.c: Topic 'channel:213038': 0x7f0c980ac890 created [Aug 18 10:33:56] DEBUG[13531] stasis.c: Creating topic. name: cache:174/channel:213038, detail: [Aug 18 10:33:56] DEBUG[13531] stasis.c: Topic 'cache:174/channel:213038': 0x7f0c980ad2c0 created [Aug 18 10:33:56] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP audio difference is 744, ms is 113 [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: Allocating new SIP dialog for 768303a53f8124c0303cd51f2f0d2f44@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13513] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c06a190' [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP allocated port 15562 [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE creating session 0.0.0.0:15562 (15562) [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE create [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE add system candidates [Aug 18 10:33:56] DEBUG[13513] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13513] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE add candidate: 159.65.48.104:15562, 2130706431 [Aug 18 10:33:56] DEBUG[13513] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13513] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE add candidate: 10.131.0.10:15562, 2130706431 [Aug 18 10:33:56] DEBUG[13513] rtp_engine.c: RTP instance '0x7f0c8c06a190' is setup and ready to go [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) ICE stopped [Aug 18 10:33:56] DEBUG[13513] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13513] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13513] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13513] res_rtp_asterisk.c: (0x7f0c8c06a190) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13513] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13513] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13513] chan_sip.c: SIP call-id changed from '768303a53f8124c0303cd51f2f0d2f44@127.0.1.1:5060' to '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13513] stasis.c: Creating topic. name: channel:213034, detail: [Aug 18 10:33:56] DEBUG[13513] stasis.c: Topic 'channel:213034': 0x7f0c8c07a490 created [Aug 18 10:33:56] DEBUG[13513] stasis.c: Creating topic. name: cache:175/channel:213034, detail: [Aug 18 10:33:56] DEBUG[13513] stasis.c: Topic 'cache:175/channel:213034': 0x7f0c8c07aec0 created [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13497] stasis/control.c: robot_213012: Adding to bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 [Aug 18 10:33:56] DEBUG[13497] stasis/app.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' is 3 interested in calls_0 [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Stopping retransmission on '710394295318048c14806fba23b501f2@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking channels create: Didn't match 213043 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:56] DEBUG[13541] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) is joining [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE create [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE add system candidates [Aug 18 10:33:56] DEBUG[13535] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13535] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE add candidate: 159.65.48.104:10382, 2130706431 [Aug 18 10:33:56] DEBUG[13535] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13535] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE add candidate: 10.131.0.10:10382, 2130706431 [Aug 18 10:33:56] DEBUG[13535] rtp_engine.c: RTP instance '0x7f0cb00e8390' is setup and ready to go [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) ICE stopped [Aug 18 10:33:56] DEBUG[13535] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13535] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13535] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13535] res_rtp_asterisk.c: (0x7f0cb00e8390) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13535] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13535] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13535] chan_sip.c: SIP call-id changed from '111105b7058ce6215acff8936326c58f@127.0.1.1:5060' to '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[13540] chan_sip.c: Audio is at 18980 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Got SDP version 812455523 and unique parts [root 812455523 IN IP4 178.62.121.41] [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 812455523 812455523 IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:56] DEBUG[13538] res_ari.c: Checking channels externalMedia: Didn't match 213043 [Aug 18 10:33:56] DEBUG[13538] res_ari.c: No explicit handler found for 213043. Using wildcard channelId. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] VERBOSE[13540] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13459] audiohook.c: Audiohook 0x7f0cac057320 has stale audio in its factories. Flushing them both [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: Allocating new SIP dialog for 4a4a54a64ac08e435646db3f6e787b5d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13534] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca40ff9e0' [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP allocated port 11300 [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE creating session 0.0.0.0:11300 (11300) [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE create [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE add system candidates [Aug 18 10:33:56] DEBUG[13534] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13534] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE add candidate: 159.65.48.104:11300, 2130706431 [Aug 18 10:33:56] DEBUG[13534] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:56] DEBUG[13534] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE add candidate: 10.131.0.10:11300, 2130706431 [Aug 18 10:33:56] DEBUG[13534] rtp_engine.c: RTP instance '0x7f0ca40ff9e0' is setup and ready to go [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) ICE stopped [Aug 18 10:33:56] DEBUG[13534] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:56] DEBUG[13534] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:56] DEBUG[13534] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:56] DEBUG[13534] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTCP setup on RTP instance [Aug 18 10:33:56] VERBOSE[13534] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:56] DEBUG[13534] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) ICE set role failed; no ice instance [Aug 18 10:33:56] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP setting address on RTP instance [Aug 18 10:33:56] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c9801eb40 -- Strict RTP learning after remote address set to: 178.62.121.41:18792 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:18792 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00f59b8) from 0x7f0c147e2330 to 0x7f0c9801b098 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00f61c8) from 0x7f0c147e2330 to 0x7f0c9801b098 [Aug 18 10:33:56] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00f6a38) from 0x7f0c147e2330 to 0x7f0c9801b098 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP ignoring duplicate property [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] VERBOSE[13540] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:56] DEBUG[13541] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: pushing 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) [Aug 18 10:33:56] DEBUG[13535] stasis.c: Creating topic. name: channel:213039, detail: [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:56] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:56] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[13183] res_rtp_asterisk.c: 0x7f0c7c0228e0 -- Strict RTP switching to RTP target address 178.62.121.41:14674 as source [Aug 18 10:33:56] DEBUG[13183] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:56] DEBUG[13183] channel.c: Channel SIP/zvonobot-0000002b setting read format path: ulaw -> alaw [Aug 18 10:33:56] DEBUG[13183] channel.c: Channel SIP/zvonobot-0000002b setting write format path: alaw -> ulaw [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13535] stasis.c: Topic 'channel:213039': 0x7f0cb00cc400 created [Aug 18 10:33:56] DEBUG[13535] stasis.c: Creating topic. name: cache:176/channel:213039, detail: [Aug 18 10:33:56] DEBUG[13535] stasis.c: Topic 'cache:176/channel:213039': 0x7f0cb00fcc50 created [Aug 18 10:33:56] VERBOSE[13540] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:56] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000030 setting read format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000030 setting write format path: alaw -> alaw [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) DTLS - ast_rtp_activate rtp=0x7f0c9801eb40 - setup and perform DTLS' [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801eb40) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801eb40) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:56] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Strict routing enforced for session 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:56] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:56] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117029@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK061e9d20 Max-Forwards: 70 From: ;tag=as08bf07d1 To: ;tag=as26319206 Contact: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] VERBOSE[12956] res_rtp_asterisk.c: 0x7f0c9c00a550 -- Strict RTP switching to RTP target address 178.62.121.41:15418 as source [Aug 18 10:33:56] DEBUG[12956] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:56] DEBUG[12956] channel.c: Channel SIP/zvonobot-0000000e setting read format path: ulaw -> alaw [Aug 18 10:33:56] DEBUG[12956] channel.c: Channel SIP/zvonobot-0000000e setting write format path: alaw -> ulaw [Aug 18 10:33:56] DEBUG[13522] res_stasis.c: calls_0: Subscribing to 213036 [Aug 18 10:33:56] DEBUG[13522] stasis/app.c: Channel '213036' is 1 interested in calls_0 [Aug 18 10:33:56] DEBUG[13522] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:56] VERBOSE[13201] dial.c: SIP/zvonobot-00000030 answered [Aug 18 10:33:56] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:56] DEBUG[13522] http.c: HTTP closing session. Top level [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Session timer started: 27 - 710394295318048c14806fba23b501f2@159.65.48.104:5060 1768000ms [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:56] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[13201] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000030 [Aug 18 10:33:56] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Outgoing Call for 79821117004 [Aug 18 10:33:56] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Initializing initreq for method INVITE - callid 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117003@178.62.121.41 SIP/2.0 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 3 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 6 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:56] DEBUG[13201] stasis/app.c: Channel '213011' is 2 interested in calls_0 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:56 GMT [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:56] VERBOSE[13541] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:56] VERBOSE[13540] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c Max-Forwards: 70 From: ;tag=as6f697fd0 To: Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 434870859 434870859 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:33:56] DEBUG[13540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:56] DEBUG[13534] chan_sip.c: SIP call-id changed from '4a4a54a64ac08e435646db3f6e787b5d@127.0.1.1:5060' to '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' [Aug 18 10:33:56] DEBUG[13534] stasis.c: Creating topic. name: channel:213041, detail: [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:56] VERBOSE[13542] chan_sip.c: Audio is at 18840 [Aug 18 10:33:56] VERBOSE[13542] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:56] VERBOSE[13540] dial.c: Called zvonobot/79821117003 [Aug 18 10:33:56] VERBOSE[13201] res_rtp_asterisk.c: 0x7f0c9801eb40 -- Strict RTP switching to RTP target address 178.62.121.41:18792 as source [Aug 18 10:33:56] DEBUG[13201] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:56] DEBUG[13201] channel.c: Channel SIP/zvonobot-00000030 setting read format path: ulaw -> alaw [Aug 18 10:33:56] DEBUG[13201] channel.c: Channel SIP/zvonobot-00000030 setting write format path: alaw -> ulaw [Aug 18 10:33:56] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 - start 1629282835.864245 answer 1629282835.914819 end 1629282836.726994 dur 0.862 bill 0.812 dispo ANSWERED [Aug 18 10:33:56] VERBOSE[13542] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] VERBOSE[13542] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c;received=159.65.48.104 From: ;tag=as6f697fd0 To: ;tag=as7ba626d2 Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="434829e8" Content-Length: 0 <-------------> [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:56] DEBUG[13474] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13504] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c;received=159.65.48.104 [Aug 18 10:33:56] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 720, ms is 65 [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13534] stasis.c: Topic 'channel:213041': 0x7f0ca410b890 created [Aug 18 10:33:56] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13536] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:56] DEBUG[13536] chan_sip.c: Allocating new SIP dialog for 35728cbb0d29ba8c36133ff211e66e77@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:56] DEBUG[13536] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac04dff0' [Aug 18 10:33:56] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP allocated port 13556 [Aug 18 10:33:56] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE creating session 0.0.0.0:13556 (13556) [Aug 18 10:33:56] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE create [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:33:56] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE add system candidates [Aug 18 10:33:56] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ba626d2 [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] DEBUG[13534] stasis.c: Creating topic. name: cache:177/channel:213041, detail: [Aug 18 10:33:56] DEBUG[13534] stasis.c: Topic 'cache:177/channel:213041': 0x7f0ca40fcd70 created [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13504] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="434829e8" [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag as7ba626d2 [Aug 18 10:33:56] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:56] DEBUG[13541] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71'. Checking compatability for channels 'Snoop/213012-00000009' and 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0' [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Initializing initreq for method INVITE - callid 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 [Aug 18 10:33:56] DEBUG[13536] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:56] DEBUG[13541] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' can not use native RTP bridge as could not get details [Aug 18 10:33:56] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 720, ms is 65 [Aug 18 10:33:56] DEBUG[13385] res_rtp_asterisk.c: (0x7f0c8c042660) RTP audio difference is 720, ms is 65 [Aug 18 10:33:56] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117004@178.62.121.41 SIP/2.0 [Aug 18 10:33:56] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13541] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:56] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP audio difference is 648, ms is 101 [Aug 18 10:33:56] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Stopping retransmission on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:56] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13541] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:56] DEBUG[13536] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d [Aug 18 10:33:56] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c Max-Forwards: 70 From: ;tag=as6f697fd0 To: ;tag=as7ba626d2 Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Audio is at 18980 [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68 Max-Forwards: 70 From: ;tag=as6f697fd0 To: Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117003@178.62.121.41", nonce="434829e8", response="cfbdda73462d5ca2bdb0f170f8a43b2c" Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 434870859 434870860 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (1) INVITE - 5 [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:33:56] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:56] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c Max-Forwards: 70 From: ;tag=as6f697fd0 To: Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 434870859 434870859 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:56] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:56] DEBUG[13541] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:56] DEBUG[13541] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:56] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Header 3 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:33:56] DEBUG[13541] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 is already using the new technology. [Aug 18 10:33:56] DEBUG[13541] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) is joining simple_bridge technology [Aug 18 10:33:56] DEBUG[13541] channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 setting read format path: slin16 -> slin16 [Aug 18 10:33:57] DEBUG[13541] channel.c: Channel Snoop/213012-00000009 setting write format path: slin16 -> slin [Aug 18 10:33:57] DEBUG[13541] channel.c: Channel Snoop/213012-00000009 setting read format path: slin -> slin16 [Aug 18 10:33:57] DEBUG[13541] channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 setting write format path: slin16 -> slin16 [Aug 18 10:33:57] DEBUG[13460] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:56] DEBUG[13542] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:56] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE add candidate: 159.65.48.104:13556, 2130706431 [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13536] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 6 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: Allocating new SIP dialog for 2416c82538b269ea46580d424b9e7fd5@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:57] DEBUG[13538] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c36530' [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) RTP allocated port 15196 [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE creating session 0.0.0.0:15196 (15196) [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE create [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE add system candidates [Aug 18 10:33:57] DEBUG[13538] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:57] DEBUG[13538] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE add candidate: 159.65.48.104:15196, 2130706431 [Aug 18 10:33:57] DEBUG[13538] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:57] DEBUG[13538] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE add candidate: 10.131.0.10:15196, 2130706431 [Aug 18 10:33:57] DEBUG[13538] rtp_engine.c: RTP instance '0x2c36530' is setup and ready to go [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) ICE stopped [Aug 18 10:33:57] DEBUG[13538] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:56 GMT [Aug 18 10:33:57] DEBUG[13538] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] VERBOSE[13542] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d Max-Forwards: 70 From: ;tag=as22c76af6 To: Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1546086222 1546086222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:33:57] DEBUG[13542] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13538] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:57] DEBUG[13459] audiohook.c: Audiohook 0x7f0cac057320 has stale audio in its factories. Flushing them both [Aug 18 10:33:57] DEBUG[13538] res_rtp_asterisk.c: (0x2c36530) RTCP setup on RTP instance [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 From: ;tag=as6f697fd0 To: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:57] VERBOSE[13538] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13536] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:57] VERBOSE[13542] dial.c: Called zvonobot/79821117004 [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag [Aug 18 10:33:57] DEBUG[13517] channel.c: Channel 0x7f0c9405cee0 'SIP/zvonobot-00000048' allocated [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13517] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13538] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13502] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #17 - INVITE (got response) [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68 Max-Forwards: 70 From: ;tag=as6f697fd0 To: Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117003@178.62.121.41", nonce="434829e8", response="cfbdda73462d5ca2bdb0f170f8a43b2c" Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 434870859 434870860 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18980 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13502] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13497] stasis/app.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' is 4 interested in calls_0 [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13545] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13545] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:57] DEBUG[13547] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13547] http.c: HTTP Request URI is /ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3/addChannel?channel=1629282835.132%2Crobot_212999 [Aug 18 10:33:57] DEBUG[13545] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13545] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13545] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13547] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13545] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13545] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13545] stasis.c: Creating topic. name: bridge:d0f9af3e-7f00-4d11-8990-3d67ba7213d6, detail: [Aug 18 10:33:57] DEBUG[13545] stasis.c: Topic 'bridge:d0f9af3e-7f00-4d11-8990-3d67ba7213d6': 0x7f0c20067620 created [Aug 18 10:33:57] DEBUG[13545] stasis.c: Creating topic. name: cache:178/bridge:d0f9af3e-7f00-4d11-8990-3d67ba7213d6, detail: [Aug 18 10:33:57] DEBUG[13545] stasis.c: Topic 'cache:178/bridge:d0f9af3e-7f00-4d11-8990-3d67ba7213d6': 0x7f0c20067290 created [Aug 18 10:33:57] DEBUG[13547] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13545] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: SIP call-id changed from '2416c82538b269ea46580d424b9e7fd5@127.0.1.1:5060' to '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' [Aug 18 10:33:57] DEBUG[13538] stasis.c: Creating topic. name: channel:213043, detail: [Aug 18 10:33:57] DEBUG[13538] stasis.c: Topic 'channel:213043': 0x2c23a70 created [Aug 18 10:33:57] DEBUG[13538] stasis.c: Creating topic. name: cache:179/channel:213043, detail: [Aug 18 10:33:57] DEBUG[13538] stasis.c: Topic 'cache:179/channel:213043': 0x2c5c060 created [Aug 18 10:33:57] DEBUG[13547] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3/addChannel] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13545] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13545] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13545] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:57] DEBUG[13545] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13545] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology constructor [Aug 18 10:33:57] DEBUG[13545] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology start [Aug 18 10:33:57] DEBUG[13545] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13545] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13547] http.c: Match made with [ari] [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 500 Server error Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c;received=159.65.48.104 From: ;tag=as6f697fd0 To: ;tag=as48575d5d Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Retry-After: 9 Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 29 instead [Aug 18 10:33:57] DEBUG[13544] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:57] DEBUG[13548] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13548] http.c: HTTP Request URI is /ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/addChannel?channel=212977 [Aug 18 10:33:57] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE add candidate: 10.131.0.10:13556, 2130706431 [Aug 18 10:33:57] DEBUG[13536] rtp_engine.c: RTP instance '0x7f0cac04dff0' is setup and ready to go [Aug 18 10:33:57] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) ICE stopped [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Finding handler for bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3/addChannel [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13548] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13548] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/addChannel] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13548] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Finding handler for bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/addChannel [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Finding handler for d0f9af3e-7f00-4d11-8990-3d67ba7213d6 [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[13548] res_ari.c: No explicit handler found for d0f9af3e-7f00-4d11-8990-3d67ba7213d6. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Finding handler for addChannel [Aug 18 10:33:57] DEBUG[13548] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:57] DEBUG[13548] stasis/control.c: 212977: Sending channel add_to_bridge command [Aug 18 10:33:57] DEBUG[13541] channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:33:57] DEBUG[13541] channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 setting write format path: slin -> slin16 [Aug 18 10:33:57] DEBUG[13536] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:57] DEBUG[13541] res_rtp_asterisk.c: (0x7f0c080871b0) RTP ooh, format changed from none to slin16 [Aug 18 10:33:57] DEBUG[13536] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:57] DEBUG[12956] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000000e [Aug 18 10:33:57] DEBUG[12956] stasis/control.c: 212977: Adding to bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 [Aug 18 10:33:57] DEBUG[13517] res_stasis.c: calls_0: Subscribing to 213035 [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13536] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:57] DEBUG[13536] res_rtp_asterisk.c: (0x7f0cac04dff0) RTCP setup on RTP instance [Aug 18 10:33:57] DEBUG[12956] stasis/app.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13517] stasis/app.c: Channel '213035' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13517] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 500 Server error [Aug 18 10:33:57] DEBUG[13517] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13550] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is joining [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Finding handler for c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK698a506c;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Outgoing Call for 79821117005 [Aug 18 10:33:57] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] VERBOSE[13536] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:57] DEBUG[13547] res_ari.c: No explicit handler found for c66c6480-4085-4bd9-87d2-ee6f5748dcc3. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:57] DEBUG[13536] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: SIP call-id changed from '35728cbb0d29ba8c36133ff211e66e77@127.0.1.1:5060' to '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' [Aug 18 10:33:57] DEBUG[13536] stasis.c: Creating topic. name: channel:213040, detail: [Aug 18 10:33:57] DEBUG[13536] stasis.c: Topic 'channel:213040': 0x7f0cac0647d0 created [Aug 18 10:33:57] DEBUG[13536] stasis.c: Creating topic. name: cache:180/channel:213040, detail: [Aug 18 10:33:57] DEBUG[13536] stasis.c: Topic 'cache:180/channel:213040': 0x7f0cac0664d0 created [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Finding handler for addChannel [Aug 18 10:33:57] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13544] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13544] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13544] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13544] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13544] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13544] stasis.c: Creating topic. name: bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060, detail: [Aug 18 10:33:57] DEBUG[13544] stasis.c: Topic 'bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060': 0x7f0c240f2ab0 created [Aug 18 10:33:57] DEBUG[13544] stasis.c: Creating topic. name: cache:181/bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060, detail: [Aug 18 10:33:57] DEBUG[13544] stasis.c: Topic 'cache:181/bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060': 0x7f0c2406c820 created [Aug 18 10:33:57] DEBUG[13547] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:33:57] DEBUG[13547] stasis/control.c: 1629282835.132: Sending channel add_to_bridge command [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13531] channel.c: Channel 0x7f0c980aab40 'SIP/zvonobot-00000049' allocated [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13531] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13531] res_stasis.c: calls_0: Subscribing to 213038 [Aug 18 10:33:57] DEBUG[13531] stasis/app.c: Channel '213038' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13531] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13531] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as48575d5d [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Outgoing Call for 79821117002 [Aug 18 10:33:57] VERBOSE[13549] chan_sip.c: Audio is at 10238 [Aug 18 10:33:57] VERBOSE[13549] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[13549] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[13549] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Initializing initreq for method INVITE - callid 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117005@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 3 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 6 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13549] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Aug 18 10:33:57] DEBUG[13549] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13474] bridge_roles.c: Roles did not exist on channel Snoop/212999-00000008 [Aug 18 10:33:57] DEBUG[13474] stasis/control.c: 1629282835.132: Adding to bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:33:57] DEBUG[13474] stasis/app.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13553] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0cb4042750(Snoop/212999-00000008) is joining [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 14]: Retry-After: 9 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag as48575d5d [Aug 18 10:33:57] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13544] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13544] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13544] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13544] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:57] DEBUG[13544] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13544] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology constructor [Aug 18 10:33:57] DEBUG[13544] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology start [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Stopping retransmission on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68 Max-Forwards: 70 From: ;tag=as6f697fd0 To: Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13544] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13550] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pushing 0x7f0c9809c220(SIP/zvonobot-0000000e) [Aug 18 10:33:57] VERBOSE[13550] bridge_channel.c: Channel SIP/zvonobot-0000000e joined 'simple_bridge' stasis-bridge [Aug 18 10:33:57] DEBUG[13544] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:33:57] VERBOSE[13549] dial.c: Called zvonobot/79821117005 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d Max-Forwards: 70 From: ;tag=as22c76af6 To: Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1546086222 1546086222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[13554] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[13552] chan_sip.c: Audio is at 17334 [Aug 18 10:33:57] VERBOSE[13552] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[13552] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[13552] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Initializing initreq for method INVITE - callid 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117002@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 3 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 6 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13552] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416056 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #37 [Aug 18 10:33:57] DEBUG[13552] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d;received=159.65.48.104 From: ;tag=as22c76af6 To: ;tag=as5c22eb4c Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75f6e2e3" Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[13550] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5c22eb4c [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75f6e2e3" [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag as5c22eb4c [Aug 18 10:33:57] DEBUG[13554] http.c: HTTP Request URI is /ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/addChannel?channel=213008 [Aug 18 10:33:57] DEBUG[13550] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:57] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP got report of 80 bytes from 178.62.121.41:11671 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Stopping retransmission on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54ba966d Max-Forwards: 70 From: ;tag=as22c76af6 To: ;tag=as5c22eb4c Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Audio is at 18840 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b Max-Forwards: 70 From: ;tag=as22c76af6 To: Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117004@178.62.121.41", nonce="75f6e2e3", response="ec9f44d9cd3689198ac34c547db27cc1" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1546086222 1546086223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #40 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 From: ;tag=as6f697fd0 To: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag [Aug 18 10:33:57] DEBUG[13550] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18;received=159.65.48.104 From: ;tag=as3cda4b3d To: ;tag=as265601b0 Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:33:57] DEBUG[13550] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13554] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13550] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:57] VERBOSE[13552] dial.c: Called zvonobot/79821117002 [Aug 18 10:33:57] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13553] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: pushing 0x7f0cb4042750(Snoop/212999-00000008) [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3cda4b3d [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as265601b0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 (Checking To) --From tag as3cda4b3d --To-tag as265601b0 [Aug 18 10:33:57] DEBUG[13554] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13554] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/addChannel] with handler [ari] len 3 [Aug 18 10:33:57] VERBOSE[13553] bridge_channel.c: Channel Snoop/212999-00000008 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:57] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 is already using the new technology. [Aug 18 10:33:57] DEBUG[13554] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is joining simple_bridge technology [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Finding handler for bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/addChannel [Aug 18 10:33:57] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Stopping retransmission on '26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:57] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13513] channel.c: Channel 0x7f0c8c078740 'SIP/zvonobot-0000004a' allocated [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13513] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Finding handler for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[13554] res_ari.c: No explicit handler found for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Finding handler for addChannel [Aug 18 10:33:57] DEBUG[13554] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:57] DEBUG[13554] stasis/control.c: 213008: Sending channel add_to_bridge command [Aug 18 10:33:57] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117042@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f965c18 Max-Forwards: 70 From: ;tag=as3cda4b3d To: ;tag=as265601b0 Contact: Call-ID: 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:57] VERBOSE[13125] dial.c: SIP/zvonobot-00000022 is busy [Aug 18 10:33:57] DEBUG[13125] channel.c: Channel 0x7f0ca80146e0 'SIP/zvonobot-00000022' hanging up. Refs: 2 [Aug 18 10:33:57] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000022 - start 1629282828.112575 answer 0.000000 end 1629282837.397000 dur 9.284 bill 1629282837.397 dispo BUSY [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13183] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000002b [Aug 18 10:33:57] DEBUG[13183] stasis/control.c: 213008: Adding to bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:33:57] DEBUG[13183] stasis/app.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:57] DEBUG[13535] channel.c: Channel 0x7f0cb00f2cd0 'SIP/zvonobot-0000004b' allocated [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13535] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13508] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 [Aug 18 10:33:57] DEBUG[13557] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13553] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13557] http.c: HTTP Request URI is /ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[13433] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 9 instead [Aug 18 10:33:57] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP changing ssrc from 1179628811 to 877211697 due to a source change [Aug 18 10:33:57] DEBUG[13553] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13553] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[12956] stasis/app.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' is 2 interested in calls_0 [Aug 18 10:33:57] DEBUG[13548] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:57] DEBUG[13556] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:33:57] DEBUG[13553] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13548] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13553] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13553] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 is already using the new technology. [Aug 18 10:33:57] DEBUG[13553] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0cb4042750(Snoop/212999-00000008) is joining simple_bridge technology [Aug 18 10:33:57] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13556] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pushing 0x7f0c7803b960(SIP/zvonobot-0000002b) [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13557] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [httpstatus] len 10 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (1) INVITE - 5 [Aug 18 10:33:57] VERBOSE[13556] bridge_channel.c: Channel SIP/zvonobot-0000002b joined 'simple_bridge' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:33:57] DEBUG[13547] stasis/control.c: robot_212999: Sending channel add_to_bridge command [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:33:57] DEBUG[13474] stasis/app.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' is 2 interested in calls_0 [Aug 18 10:33:57] DEBUG[13558] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:33:57] DEBUG[13557] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13557] http.c: match request [ari/bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13557] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Finding handler for bridges/0aaea81d-67a8-499e-9e08-2fb745e40804/play [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Finding handler for 0aaea81d-67a8-499e-9e08-2fb745e40804 [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[13557] res_ari.c: No explicit handler found for 0aaea81d-67a8-499e-9e08-2fb745e40804. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Finding handler for play [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:57] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:57] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13558] http.c: HTTP Request URI is /ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/record?name=212977_vtgwVPwmpVRoqoJmilBCHrdNgFSFbIMS&format=wav [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:57] DEBUG[13557] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416056 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 13 instead [Aug 18 10:33:57] DEBUG[13556] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13556] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13556] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13556] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13556] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13556] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 is already using the new technology. [Aug 18 10:33:57] DEBUG[13556] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining simple_bridge technology [Aug 18 10:33:57] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13535] res_stasis.c: calls_0: Subscribing to 213039 [Aug 18 10:33:57] DEBUG[13535] stasis/app.c: Channel '213039' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13535] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13558] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/record] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13535] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13513] res_stasis.c: calls_0: Subscribing to 213034 [Aug 18 10:33:57] DEBUG[13513] stasis/app.c: Channel '213034' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13513] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP changing ssrc from 124083079 to 1443903994 due to a source change [Aug 18 10:33:57] DEBUG[13513] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13554] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:57] DEBUG[13554] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Outgoing Call for 79821117006 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Outgoing Call for 79821117001 [Aug 18 10:33:57] DEBUG[13562] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13183] stasis/app.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' is 2 interested in calls_0 [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13558] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/record] with handler [phoneprov] len 9 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423 Max-Forwards: 70 From: ;tag=as46ab0e55 To: ;tag=as396a139d Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="39509751", response="ab8220e1f7ce5805733f6c6a4eac6fb4" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as46ab0e55 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as396a139d [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="39509751", response="ab8220e1f7ce5805733f6c6a4eac6fb4" [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 (Checking From) --From tag as46ab0e55 --To-tag as396a139d [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:57] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: Allocating new SIP dialog for 2cb40b15599b80897aad6f6656948c1e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:57] DEBUG[13534] channel.c: Channel 0x7f0ca41093f0 'SIP/zvonobot-0000004c' allocated [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13534] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13558] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/record] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[13559] chan_sip.c: Audio is at 10382 [Aug 18 10:33:57] VERBOSE[13559] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[13559] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] DEBUG[13539] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c100e4670' [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) RTP allocated port 11140 [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE creating session 0.0.0.0:11140 (11140) [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE create [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE add system candidates [Aug 18 10:33:57] DEBUG[13534] res_stasis.c: calls_0: Subscribing to 213041 [Aug 18 10:33:57] DEBUG[13534] stasis/app.c: Channel '213041' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13539] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:57] DEBUG[13539] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE add candidate: 159.65.48.104:11140, 2130706431 [Aug 18 10:33:57] DEBUG[13539] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:57] DEBUG[13539] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:57] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13534] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13534] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13562] http.c: HTTP Request URI is /ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/record?name=213008_uRBdlcrkyUhysKyMPjaNFREMlcwBMbYe&format=wav [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[13558] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Finding handler for bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Finding handler for d0f9af3e-7f00-4d11-8990-3d67ba7213d6 [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[13558] res_ari.c: No explicit handler found for d0f9af3e-7f00-4d11-8990-3d67ba7213d6. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Finding handler for record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:57] DEBUG[13558] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:57] DEBUG[13558] stasis.c: Creating topic. name: channel:1629282837.153, detail: [Aug 18 10:33:57] DEBUG[13558] stasis.c: Topic 'channel:1629282837.153': 0x7f0c7c02e7e0 created [Aug 18 10:33:57] DEBUG[13558] stasis.c: Creating topic. name: cache:182/channel:1629282837.153, detail: [Aug 18 10:33:57] DEBUG[13558] stasis.c: Topic 'cache:182/channel:1629282837.153': 0x7f0c7c07cf30 created [Aug 18 10:33:57] VERBOSE[13559] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Initializing initreq for method INVITE - callid 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117001@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 3 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE add candidate: 10.131.0.10:11140, 2130706431 [Aug 18 10:33:57] DEBUG[13539] rtp_engine.c: RTP instance '0x7f0c100e4670' is setup and ready to go [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) ICE stopped [Aug 18 10:33:57] DEBUG[13539] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:57] DEBUG[13539] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:57] DEBUG[13539] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:57] DEBUG[13539] res_rtp_asterisk.c: (0x7f0c100e4670) RTCP setup on RTP instance [Aug 18 10:33:57] VERBOSE[13539] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Outgoing Call for 79821116999 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[13562] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/record] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[13561] chan_sip.c: Audio is at 15562 [Aug 18 10:33:57] VERBOSE[13561] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:57] DEBUG[13294] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-0000001b' [Aug 18 10:33:57] DEBUG[13539] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: SIP call-id changed from '2cb40b15599b80897aad6f6656948c1e@127.0.1.1:5060' to '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' [Aug 18 10:33:57] DEBUG[13539] stasis.c: Creating topic. name: channel:213042, detail: [Aug 18 10:33:57] DEBUG[13539] stasis.c: Topic 'channel:213042': 0x7f0c100f05f0 created [Aug 18 10:33:57] DEBUG[13539] stasis.c: Creating topic. name: cache:183/channel:213042, detail: [Aug 18 10:33:57] DEBUG[13539] stasis.c: Topic 'cache:183/channel:213042': 0x7f0c100f1070 created [Aug 18 10:33:57] VERBOSE[13561] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 880, ms is 75 [Aug 18 10:33:57] DEBUG[13562] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/record] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 6 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13504] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13562] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/record] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13562] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13559] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437933 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Aug 18 10:33:57] DEBUG[13559] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Finding handler for bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/record [Aug 18 10:33:57] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] DEBUG[13294] res_rtp_asterisk.c: (0x7f0c8001c6f0) RTP no remote address on instance, so dropping frame [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[13508] stasis/control.c: robot_212999: Adding to bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13508] stasis/app.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' is 3 interested in calls_0 [Aug 18 10:33:57] VERBOSE[13561] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Initializing initreq for method INVITE - callid 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117006@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 3 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 6 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13561] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #48 [Aug 18 10:33:57] DEBUG[13561] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] VERBOSE[13559] dial.c: Called zvonobot/79821117001 [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Finding handler for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:57] DEBUG[13562] res_ari.c: No explicit handler found for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060. Using wildcard bridgeId. [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Finding handler for record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:57] DEBUG[13562] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:57] DEBUG[13562] stasis.c: Creating topic. name: channel:1629282837.155, detail: [Aug 18 10:33:57] DEBUG[13562] stasis.c: Topic 'channel:1629282837.155': 0x7f0c80052540 created [Aug 18 10:33:57] DEBUG[13562] stasis.c: Creating topic. name: cache:184/channel:1629282837.155, detail: [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13562] stasis.c: Topic 'cache:184/channel:1629282837.155': 0x7f0c80052f90 created [Aug 18 10:33:57] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13564] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) is joining [Aug 18 10:33:57] DEBUG[13294] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-0000001b' [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] VERBOSE[13561] dial.c: Called zvonobot/79821117006 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423;received=178.62.121.41 From: ;tag=as46ab0e55 To: ;tag=as396a139d Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Setting 0x7f0c8c00b190(SIP/zvonobot-0000001b) state from:0 to:1 [Aug 18 10:33:57] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pulling 0x7f0c8c00b190(SIP/zvonobot-0000001b) [Aug 18 10:33:57] VERBOSE[13294] bridge_channel.c: Channel SIP/zvonobot-0000001b left 'softmix' stasis-bridge [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[13564] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: pushing 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 9 instead [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c8c00b190(SIP/zvonobot-0000001b) is leaving softmix technology [Aug 18 10:33:57] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13499] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13460] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:57] DEBUG[13460] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[13460] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Setting 0x7f0c7804b4e0(Announcer/ARI-00000010;2) state from:0 to:2 [Aug 18 10:33:57] VERBOSE[13563] chan_sip.c: Audio is at 11300 [Aug 18 10:33:57] VERBOSE[13564] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:33:57] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50116-0x7f0c24077280 - start 1629282836.007844 answer 1629282836.105462 end 1629282837.707378 dur 1.699 bill 1.601 dispo ANSWERED [Aug 18 10:33:57] DEBUG[13460] channel.c: Channel Announcer/ARI-00000012;1 setting write format path: slin -> slin [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b Max-Forwards: 70 From: ;tag=as22c76af6 To: Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117004@178.62.121.41", nonce="75f6e2e3", response="ec9f44d9cd3689198ac34c547db27cc1" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1546086222 1546086223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (2) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] VERBOSE[13563] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[13563] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[13563] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Initializing initreq for method INVITE - callid 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116999@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 3 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 6 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13563] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801089 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Aug 18 10:33:57] DEBUG[13563] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (2) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416056 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Session timer stopped: 9 - 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa;received=159.65.48.104 From: ;tag=as30c9fc1b To: ;tag=as1917d2c9 Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17c0a9ea" Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1917d2c9 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17c0a9ea" [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag as1917d2c9 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Aug 18 10:33:57] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13460] channel.c: Channel 0x7f0c3009e0e0 'Announcer/ARI-00000012;1' hanging up. Refs: 2 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Stopping retransmission on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:33:57] VERBOSE[13563] dial.c: Called zvonobot/79821116999 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa Max-Forwards: 70 From: ;tag=as30c9fc1b To: ;tag=as1917d2c9 Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:57] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13294] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59'. Checking compatability for channels 'Announcer/ARI-00000010;2' and 'Recorder/ARI-00000007;2' [Aug 18 10:33:57] DEBUG[13294] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as could not get details [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13294] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] VERBOSE[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: switching from softmix technology to simple_bridge [Aug 18 10:33:57] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001b - start 1629282826.190972 answer 1629282832.216468 end 1629282837.747618 dur 11.556 bill 5.531 dispo ANSWERED [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology constructor [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c7804b4e0(Announcer/ARI-00000010;2) to dummy bridge temporarily [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13564] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3'. Checking compatability for channels 'Snoop/212999-00000008' and 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280' [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Audio is at 10238 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033 Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117005@178.62.121.41", nonce="17c0a9ea", response="7c1cb2cc0cbfa4cd0b231dd890a72a4f" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: moving 0x7f0c9802d570(Recorder/ARI-00000007;2) to dummy bridge temporarily [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c7804b4e0(Announcer/ARI-00000010;2) is leaving softmix technology (dummy) [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is leaving softmix technology (dummy) [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology stop [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c7804b4e0(Announcer/ARI-00000010;2) is joining simple_bridge technology [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Recorder/ARI-00000007;2 setting read format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Announcer/ARI-00000010;2 setting write format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Announcer/ARI-00000010;2 setting read format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is joining simple_bridge technology [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Recorder/ARI-00000007;2 setting read format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Announcer/ARI-00000010;2 setting write format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Announcer/ARI-00000010;2 setting read format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel Recorder/ARI-00000007;2 setting write format path: slin -> slin [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology start [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437933 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: deferring softmix technology destructor [Aug 18 10:33:57] DEBUG[13564] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' can not use native RTP bridge as could not get details [Aug 18 10:33:57] DEBUG[13294] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: queueing action type:13 sub:1000 [Aug 18 10:33:57] DEBUG[13538] channel.c: Channel 0x2c3ac80 'SIP/zvonobot-0000004d' allocated [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (2) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117004@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b Max-Forwards: 70 From: ;tag=as22c76af6 To: Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117004@178.62.121.41", nonce="75f6e2e3", response="ec9f44d9cd3689198ac34c547db27cc1" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1546086222 1546086223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13564] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13564] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13538] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13538] res_stasis.c: calls_0: Subscribing to 213043 [Aug 18 10:33:57] DEBUG[13538] stasis/app.c: Channel '213043' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13538] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13538] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13564] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13564] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13564] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 is already using the new technology. [Aug 18 10:33:57] DEBUG[13564] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) is joining simple_bridge technology [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3;received=159.65.48.104 From: ;tag=as19ef62c2 To: ;tag=as062fa867 Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3423c0a6" Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 setting read format path: slin16 -> slin16 [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel Snoop/212999-00000008 setting write format path: slin16 -> slin [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel Snoop/212999-00000008 setting read format path: slin -> slin16 [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 setting write format path: slin16 -> slin16 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as062fa867 [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Outgoing Call for 79821116997 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:57] DEBUG[13125] chan_sip.c: Hangup call SIP/zvonobot-00000022, SIP callid 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13125] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] DEBUG[13125] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:57] DEBUG[13536] channel.c: Channel 0x7f0cac053540 'SIP/zvonobot-0000004e' allocated [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13536] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13125] channel.c: Channel 0x7f0ca80146e0 'SIP/zvonobot-00000022' destroying [Aug 18 10:33:57] DEBUG[13558] channel.c: Channel 0x7f0c7c07a200 'Recorder/ARI-00000019;1' allocated [Aug 18 10:33:57] DEBUG[13558] stasis.c: Creating topic. name: channel:1629282837.156, detail: [Aug 18 10:33:57] DEBUG[13558] stasis.c: Topic 'channel:1629282837.156': 0x7f0c7c07c9c0 created [Aug 18 10:33:57] DEBUG[13558] stasis.c: Creating topic. name: cache:185/channel:1629282837.156, detail: [Aug 18 10:33:57] DEBUG[13558] stasis.c: Topic 'cache:185/channel:1629282837.156': 0x7f0c7c01b510 created [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3423c0a6" [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag as062fa867 [Aug 18 10:33:57] DEBUG[13536] res_stasis.c: calls_0: Subscribing to 213040 [Aug 18 10:33:57] DEBUG[13536] stasis/app.c: Channel '213040' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13536] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Outgoing Call for 79821117000 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:57] DEBUG[13536] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13432] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: stopping mixing thread [Aug 18 10:33:57] DEBUG[13431] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pulling 0x7f0c7804b4e0(Announcer/ARI-00000010;2) [Aug 18 10:33:57] VERBOSE[13431] bridge_channel.c: Channel Announcer/ARI-00000010;2 left 'simple_bridge' stasis-bridge [Aug 18 10:33:57] DEBUG[13431] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c7804b4e0(Announcer/ARI-00000010;2) is leaving simple_bridge technology [Aug 18 10:33:57] DEBUG[13431] bridge_channel.c: Setting 0x7f0c9802d570(Recorder/ARI-00000007;2) state from:0 to:2 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:57] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:57] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[13568] chan_sip.c: Audio is at 15196 [Aug 18 10:33:57] VERBOSE[13568] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[13568] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[13568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Initializing initreq for method INVITE - callid 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116997@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 3 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 6 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13568] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Aug 18 10:33:57] DEBUG[13568] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:33:57] DEBUG[20620] stasis/app.c: channel '212998': is 0 interested in calls_0 [Aug 18 10:33:57] DEBUG[20620] stasis/app.c: channel '212998' unsubscribed from calls_0 [Aug 18 10:33:57] DEBUG[20620] stasis.c: Destroying topic. name: cache:50/channel:212998, detail: [Aug 18 10:33:57] DEBUG[20620] stasis.c: Topic 'cache:50/channel:212998': 0x7f0ca80167f0 destroyed [Aug 18 10:33:57] DEBUG[20620] stasis.c: Destroying topic. name: channel:212998, detail: [Aug 18 10:33:57] DEBUG[20620] stasis.c: Topic 'channel:212998': 0x7f0ca8016710 destroyed [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:57] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282837.157, detail: [Aug 18 10:33:57] DEBUG[20545] stasis.c: Topic 'channel:1629282837.157': 0x7f0c300d8eb0 created [Aug 18 10:33:57] DEBUG[20545] stasis.c: Creating topic. name: cache:186/channel:1629282837.157, detail: [Aug 18 10:33:57] DEBUG[20545] stasis.c: Topic 'cache:186/channel:1629282837.157': 0x7f0c300b3540 created [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Stopping retransmission on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:57] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[20545] stasis.c: Destroying topic. name: cache:186/channel:1629282837.157, detail: [Aug 18 10:33:57] DEBUG[20545] stasis.c: Topic 'cache:186/channel:1629282837.157': 0x7f0c300b3540 destroyed [Aug 18 10:33:57] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282837.157, detail: [Aug 18 10:33:57] DEBUG[20545] stasis.c: Topic 'channel:1629282837.157': 0x7f0c300d8eb0 destroyed [Aug 18 10:33:57] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:48', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000022', '', 'AppDial2', '(Outgoing Line)', 9, 0, 'BUSY', 3, '', '212998', '')] [Aug 18 10:33:57] DEBUG[13431] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13431] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13431] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13431] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] VERBOSE[13568] dial.c: Called zvonobot/79821116997 [Aug 18 10:33:57] DEBUG[13431] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] DEBUG[13431] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 is already using the new technology. [Aug 18 10:33:57] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] VERBOSE[13569] chan_sip.c: Audio is at 13556 [Aug 18 10:33:57] VERBOSE[13569] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:57] DEBUG[13298] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: pulling 0x7f0c9802d570(Recorder/ARI-00000007;2) [Aug 18 10:33:57] DEBUG[20534] bridge_softmix.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: Waiting for mixing thread to die. [Aug 18 10:33:57] VERBOSE[13298] bridge_channel.c: Channel Recorder/ARI-00000007;2 left 'simple_bridge' stasis-bridge [Aug 18 10:33:57] DEBUG[13298] bridge_channel.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: 0x7f0c9802d570(Recorder/ARI-00000007;2) is leaving simple_bridge technology [Aug 18 10:33:57] DEBUG[13298] bridge_native_rtp.c: Bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' can not use native RTP bridge as two channels are required [Aug 18 10:33:57] DEBUG[13298] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:57] DEBUG[13298] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13298] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:57] DEBUG[13298] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:57] DEBUG[13298] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 is already using the new technology. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:57] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Bridge is returning 0x7f0c8c00b190(SIP/zvonobot-0000001b) to read format alaw [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3 Max-Forwards: 70 From: ;tag=as19ef62c2 To: ;tag=as062fa867 Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel SIP/zvonobot-0000001b setting read format path: alaw -> alaw [Aug 18 10:33:57] DEBUG[13431] channel.c: Channel 0x7f0c78050c90 'Announcer/ARI-00000010;2' hanging up. Refs: 2 [Aug 18 10:33:57] DEBUG[13298] channel.c: Channel 0x7f0c980450a0 'Recorder/ARI-00000007;2' hanging up. Refs: 2 [Aug 18 10:33:57] DEBUG[13294] bridge_channel.c: Bridge is returning 0x7f0c8c00b190(SIP/zvonobot-0000001b) to write format alaw [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:57] VERBOSE[13569] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] DEBUG[13294] channel.c: Channel SIP/zvonobot-0000001b setting write format path: alaw -> alaw [Aug 18 10:33:57] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:57] DEBUG[13294] stasis/control.c: 212993, a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: Channel was departed from bridge [Aug 18 10:33:57] DEBUG[13294] stasis/app.c: bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13044] stasis/control.c: 212993: Channel departing bridge [Aug 18 10:33:57] DEBUG[13044] bridge.c: Waiting for 0x7f0c8c00b190(SIP/zvonobot-0000001b) bridge thread to die. [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Audio is at 17334 [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117002@178.62.121.41", nonce="3423c0a6", response="6b620ac742be9e83a3506227f8751ecd" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416057 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13294] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:33:57] VERBOSE[13569] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:57] DEBUG[13044] stasis/app.c: channel '212993': is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13508] stasis/app.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' is 4 interested in calls_0 [Aug 18 10:33:57] DEBUG[13570] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12 instead [Aug 18 10:33:57] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13547] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:57] DEBUG[13547] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13044] channel.c: Channel 0x7f0c800219b0 'SIP/zvonobot-0000001b' hanging up. Refs: 3 [Aug 18 10:33:57] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 656, ms is 61 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Initializing initreq for method INVITE - callid 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 0 [ 44]: INVITE sip:79821117000@178.62.121.41 SIP/2.0 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 3 [ 52]: From: ;tag=as6093d024 [Aug 18 10:33:57] DEBUG[13570] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:57] DEBUG[13570] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13570] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13570] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13570] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13570] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801089 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (1) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033 Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117005@178.62.121.41", nonce="17c0a9ea", response="7c1cb2cc0cbfa4cd0b231dd890a72a4f" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (2) INVITE - 5 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437933 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 6 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:57 GMT [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:57] VERBOSE[13569] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160452 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #47 [Aug 18 10:33:57] DEBUG[13569] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:57] DEBUG[13573] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13573] http.c: HTTP Request URI is /ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:57] DEBUG[13573] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13573] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13573] http.c: match request [ari/bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13573] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Finding handler for bridges/e2e70698-2279-429d-a48c-2fe9dd817267/play [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Finding handler for bridges [Aug 18 10:33:57] DEBUG[13584] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13584] http.c: HTTP Request URI is /ari/channels/212998 [Aug 18 10:33:57] DEBUG[13570] stasis.c: Creating topic. name: bridge:e594e1d1-53fe-4904-8517-472d8e3b8b52, detail: [Aug 18 10:33:57] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13584] http.c: match request [ari/channels/212998] with handler [httpstatus] len 10 [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:57] DEBUG[13584] http.c: match request [ari/channels/212998] with handler [phoneprov] len 9 [Aug 18 10:33:57] DEBUG[13584] http.c: match request [ari/channels/212998] with handler [ari] len 3 [Aug 18 10:33:57] DEBUG[13584] http.c: Match made with [ari] [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Finding handler for channels/212998 [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:33:57] DEBUG[13570] stasis.c: Topic 'bridge:e594e1d1-53fe-4904-8517-472d8e3b8b52': 0x7f0ca8009cd0 created [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:57] VERBOSE[13569] dial.c: Called zvonobot/79821117000 [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Finding handler for channels [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:57] DEBUG[13585] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:57] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 From: ;tag=as22c76af6 To: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:33:57] DEBUG[13564] channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 setting write format path: slin -> slin16 [Aug 18 10:33:57] DEBUG[13564] res_rtp_asterisk.c: (0x7f0c24077280) RTP ooh, format changed from none to slin16 [Aug 18 10:33:57] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:57] DEBUG[13539] channel.c: Channel 0x7f0c100ee3b0 'SIP/zvonobot-0000004f' allocated [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:57] DEBUG[13539] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:57] DEBUG[13539] res_stasis.c: calls_0: Subscribing to 213042 [Aug 18 10:33:57] DEBUG[13539] stasis/app.c: Channel '213042' is 1 interested in calls_0 [Aug 18 10:33:57] DEBUG[13539] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Finding handler for 212998 [Aug 18 10:33:57] DEBUG[13539] http.c: HTTP closing session. Top level [Aug 18 10:33:57] DEBUG[13570] stasis.c: Creating topic. name: cache:187/bridge:e594e1d1-53fe-4904-8517-472d8e3b8b52, detail: [Aug 18 10:33:57] DEBUG[13573] res_ari.c: Finding handler for e2e70698-2279-429d-a48c-2fe9dd817267 [Aug 18 10:33:57] DEBUG[13460] channel.c: Channel 0x7f0c3009e0e0 'Announcer/ARI-00000012;1' destroying [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:57] DEBUG[13562] channel.c: Channel 0x7f0c800507f0 'Recorder/ARI-0000001a;1' allocated [Aug 18 10:33:57] DEBUG[13585] http.c: HTTP Request URI is /ari/playbacks/71191322-0703-4b74-a621-247adf7188a9 [Aug 18 10:33:57] DEBUG[13570] stasis.c: Topic 'cache:187/bridge:e594e1d1-53fe-4904-8517-472d8e3b8b52': 0x7f0ca80433a0 created [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking channels create: Didn't match 212998 [Aug 18 10:33:57] DEBUG[13562] stasis.c: Creating topic. name: channel:1629282837.158, detail: [Aug 18 10:33:57] DEBUG[13455] bridge_channel.c: Setting 0x7f0c300a4c90(Announcer/ARI-00000012;2) state from:0 to:1 [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13573] res_ari.c: No explicit handler found for e2e70698-2279-429d-a48c-2fe9dd817267. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Finding handler for play [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:58] DEBUG[13573] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:57] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:58] DEBUG[13562] stasis.c: Topic 'channel:1629282837.158': 0x7f0c800526e0 created [Aug 18 10:33:58] DEBUG[13562] stasis.c: Creating topic. name: cache:188/channel:1629282837.158, detail: [Aug 18 10:33:58] DEBUG[13562] stasis.c: Topic 'cache:188/channel:1629282837.158': 0x7f0c80052830 created [Aug 18 10:33:58] DEBUG[13589] http.c: HTTP opening session. Top level [Aug 18 10:33:57] DEBUG[13584] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13584] res_ari.c: Checking channels externalMedia: Didn't match 212998 [Aug 18 10:33:58] DEBUG[13584] res_ari.c: No explicit handler found for 212998. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13589] http.c: HTTP Request URI is /ari/channels/213044?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116996&callerId=74950493843 [Aug 18 10:33:57] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[13455] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pulling 0x7f0c300a4c90(Announcer/ARI-00000012;2) [Aug 18 10:33:58] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] VERBOSE[13455] bridge_channel.c: Channel Announcer/ARI-00000012;2 left 'softmix' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:33:58] DEBUG[13455] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c300a4c90(Announcer/ARI-00000012;2) is leaving softmix technology [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #40 - INVITE (got response) [Aug 18 10:33:58] DEBUG[13585] http.c: match request [ari/playbacks/71191322-0703-4b74-a621-247adf7188a9] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13570] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13570] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13592] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13589] http.c: match request [ari/channels/213044] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13570] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13585] http.c: match request [ari/playbacks/71191322-0703-4b74-a621-247adf7188a9] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:58] DEBUG[13460] stasis.c: Destroying topic. name: cache:150/channel:1629282835.125, detail: [Aug 18 10:33:58] DEBUG[13460] stasis.c: Topic 'cache:150/channel:1629282835.125': 0x7f0c3007c870 destroyed [Aug 18 10:33:58] DEBUG[13460] stasis.c: Destroying topic. name: channel:1629282835.125, detail: [Aug 18 10:33:58] DEBUG[13460] stasis.c: Topic 'channel:1629282835.125': 0x7f0c30071040 destroyed [Aug 18 10:33:58] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:58] DEBUG[13455] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804'. Checking compatability for channels 'SIP/zvonobot-00000020' and 'Recorder/ARI-00000011;2' [Aug 18 10:33:58] DEBUG[13455] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as channel 'SIP/zvonobot-00000020' has features which prevent it [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13455] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] VERBOSE[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: switching from softmix technology to simple_bridge [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology constructor [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c88050c90(SIP/zvonobot-00000020) to dummy bridge temporarily [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c180a6470(Recorder/ARI-00000011;2) to dummy bridge temporarily [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is leaving softmix technology (dummy) [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is leaving softmix technology (dummy) [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology stop [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel Recorder/ARI-00000011;2 setting read format path: slin -> slin [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel Recorder/ARI-00000011;2 setting read format path: slin -> slin [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology start [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: deferring softmix technology destructor [Aug 18 10:33:58] DEBUG[13455] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: queueing action type:13 sub:1000 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13592] http.c: HTTP Request URI is /ari/channels/213046?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116994&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13592] http.c: match request [ari/channels/213046] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13592] http.c: match request [ari/channels/213046] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13592] http.c: match request [ari/channels/213046] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13592] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13570] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:58] DEBUG[13589] http.c: match request [ari/channels/213044] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[13594] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13570] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Outgoing Call for 79821116998 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[13587] chan_sip.c: Audio is at 11140 [Aug 18 10:33:58] VERBOSE[13587] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[13587] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[13587] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Initializing initreq for method INVITE - callid 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116998@178.62.121.41 SIP/2.0 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 3 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 6 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13570] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology constructor [Aug 18 10:33:58] DEBUG[13570] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology start [Aug 18 10:33:58] DEBUG[13594] http.c: HTTP Request URI is /ari/channels/213048?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116992&callerId=74950493843 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:33:58] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Destroying SIP dialog 26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '26ae6b4c5c782b5a0b25aed30bdf492d@159.65.48.104:5060' Method: INVITE [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS stop [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800f800) ICE RTP transport deallocating [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca800f800' [Aug 18 10:33:58] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13589] http.c: match request [ari/channels/213044] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13594] http.c: match request [ari/channels/213048] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[13385] res_rtp_asterisk.c: (0x7f0c8c042660) RTP audio difference is 784, ms is 69 [Aug 18 10:33:58] DEBUG[13589] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13585] http.c: match request [ari/playbacks/71191322-0703-4b74-a621-247adf7188a9] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13585] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Finding handler for playbacks/71191322-0703-4b74-a621-247adf7188a9 [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Finding handler for playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Finding handler for 71191322-0703-4b74-a621-247adf7188a9 [Aug 18 10:33:58] DEBUG[13585] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13585] res_ari.c: No explicit handler found for 71191322-0703-4b74-a621-247adf7188a9. Using wildcard playbackId. [Aug 18 10:33:58] DEBUG[13585] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:58] DEBUG[13606] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13594] http.c: match request [ari/channels/213048] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:58] DEBUG[13594] http.c: match request [ari/channels/213048] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13594] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13594] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Finding handler for channels/213048 [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Finding handler for 213048 [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking channels create: Didn't match 213048 [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13594] res_ari.c: Checking channels externalMedia: Didn't match 213048 [Aug 18 10:33:58] DEBUG[13594] res_ari.c: No explicit handler found for 213048. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13592] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13602] http.c: HTTP opening session. Top level [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117002@178.62.121.41", nonce="3423c0a6", response="6b620ac742be9e83a3506227f8751ecd" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416057 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[13585] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801089 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13433] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13433] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13433] channel.c: Channel Announcer/ARI-00000010;1 setting write format path: slin -> slin [Aug 18 10:33:58] NOTICE[13433] res_stasis_playback.c: 1629282834.120: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13606] http.c: HTTP Request URI is /ari/channels/213047?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116993&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13602] http.c: HTTP Request URI is /ari/channels/213051?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116989&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13608] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13589] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13456] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: stopping mixing thread [Aug 18 10:33:58] DEBUG[20534] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:58] DEBUG[13602] http.c: match request [ari/channels/213051] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13570] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13608] http.c: HTTP Request URI is /ari/channels/robot_212993 [Aug 18 10:33:58] DEBUG[13608] http.c: match request [ari/channels/robot_212993] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13570] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13606] http.c: match request [ari/channels/213047] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13602] http.c: match request [ari/channels/213051] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[20534] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: Waiting for mixing thread to die. [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #47 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #47)) [Aug 18 10:33:58] DEBUG[13433] channel.c: Channel 0x7f0c780058a0 'Announcer/ARI-00000010;1' hanging up. Refs: 2 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:33:58] DEBUG[13443] channel.c: Recorder/ARI-00000011;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160452 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13606] http.c: match request [ari/channels/213047] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13609] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Finding handler for channels/213046 [Aug 18 10:33:58] DEBUG[13606] http.c: match request [ari/channels/213047] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Finding handler for channels/213044 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13610] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13606] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13602] http.c: match request [ari/channels/213051] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13606] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Finding handler for channels/213047 [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Finding handler for 213047 [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking channels create: Didn't match 213047 [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13606] res_ari.c: Checking channels externalMedia: Didn't match 213047 [Aug 18 10:33:58] DEBUG[13606] res_ari.c: No explicit handler found for 213047. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13602] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:58 GMT [Aug 18 10:33:58] DEBUG[13608] http.c: match request [ari/channels/robot_212993] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel 0x7f0c300a36a0 'Announcer/ARI-00000012;2' hanging up. Refs: 2 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13602] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13613] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Finding handler for channels/213051 [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:58] VERBOSE[13587] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010121 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #41 [Aug 18 10:33:58] DEBUG[13587] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13609] http.c: HTTP Request URI is /ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/addChannel?channel=213011 [Aug 18 10:33:58] DEBUG[13611] http.c: HTTP opening session. Top level [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 895903859 895903859 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10224 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4f7b8e6e [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as121e4580 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[13441] channel.c: SIP/zvonobot-00000020: Dropping redundant connected line update "" <>. [Aug 18 10:33:58] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 895903859 895903859 IN IP4 178.62.121.41 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10224 RTP/AVP 0 8 101 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 (Checking To) --From tag as4f7b8e6e --To-tag as121e4580 [Aug 18 10:33:58] DEBUG[13613] http.c: HTTP Request URI is /ari/channels/213052?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116988&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: Allocating new SIP dialog for 20a251b0721862b6130f69bd013031b8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13606] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c100f9ed0' [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) RTP allocated port 11378 [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE creating session 0.0.0.0:11378 (11378) [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE create [Aug 18 10:33:58] DEBUG[13608] http.c: match request [ari/channels/robot_212993] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13609] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] VERBOSE[13327] res_rtp_asterisk.c: 0x7f0cb003cb10 -- Strict RTP learning complete - Locking on source address 178.62.121.41:12912 [Aug 18 10:33:58] DEBUG[13610] http.c: HTTP Request URI is /ari/channels/213045?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116995&callerId=74950493843 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Finding handler for 213046 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking channels create: Didn't match 213046 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13592] res_ari.c: Checking channels externalMedia: Didn't match 213046 [Aug 18 10:33:58] DEBUG[13592] res_ari.c: No explicit handler found for 213046. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13611] http.c: HTTP Request URI is /ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:58] DEBUG[13608] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Finding handler for 213044 [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking channels create: Didn't match 213044 [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13609] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13609] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/addChannel] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13614] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13589] res_ari.c: Checking channels externalMedia: Didn't match 213044 [Aug 18 10:33:58] DEBUG[13610] http.c: match request [ari/channels/213045] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13614] http.c: HTTP Request URI is /ari/channels/213053?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116987&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13614] http.c: match request [ari/channels/213053] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE add system candidates [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Got SDP version 895903859 and unique parts [root 895903859 IN IP4 178.62.121.41] [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 895903859 895903859 IN IP4 178.62.121.41... OK. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:58] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:58] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c01e650) ICE set role failed; no ice instance [Aug 18 10:33:58] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP setting address on RTP instance [Aug 18 10:33:58] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c7c037ee0 -- Strict RTP learning after remote address set to: 178.62.121.41:10224 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10224 [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0067e18) from 0x7f0c147e2330 to 0x7f0c7c01e828 [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00e4748) from 0x7f0c147e2330 to 0x7f0c7c01e828 [Aug 18 10:33:58] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0107788) from 0x7f0c147e2330 to 0x7f0c7c01e828 [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP ignoring duplicate property [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:58] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003b setting read format path: alaw -> alaw [Aug 18 10:33:58] DEBUG[13606] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13606] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE add candidate: 159.65.48.104:11378, 2130706431 [Aug 18 10:33:58] DEBUG[13606] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13606] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE add candidate: 10.131.0.10:11378, 2130706431 [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Finding handler for channels/robot_212993 [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13609] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13589] res_ari.c: No explicit handler found for 213044. Using wildcard channelId. [Aug 18 10:33:58] VERBOSE[13587] dial.c: Called zvonobot/79821116998 [Aug 18 10:33:58] DEBUG[13614] http.c: match request [ari/channels/213053] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13614] http.c: match request [ari/channels/213053] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13614] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: Allocating new SIP dialog for 2a995dc63644d08a3c6298b03118c0d8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13594] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4045900' [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) RTP allocated port 18824 [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE creating session 0.0.0.0:18824 (18824) [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE create [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add system candidates [Aug 18 10:33:58] DEBUG[13594] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13594] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add candidate: 159.65.48.104:18824, 2130706431 [Aug 18 10:33:58] DEBUG[13594] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13594] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add candidate: 10.131.0.10:18824, 2130706431 [Aug 18 10:33:58] DEBUG[13594] rtp_engine.c: RTP instance '0x7f0cb4045900' is setup and ready to go [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) ICE stopped [Aug 18 10:33:58] DEBUG[13594] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13594] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13594] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13594] res_rtp_asterisk.c: (0x7f0cb4045900) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13594] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13594] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: SIP call-id changed from '2a995dc63644d08a3c6298b03118c0d8@127.0.1.1:5060' to '596ef69a675656833620d8345b47f33c@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003b setting write format path: alaw -> alaw [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c01e650) DTLS - ast_rtp_activate rtp=0x7f0c7c037ee0 - setup and perform DTLS' [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c037ee0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c037ee0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:58] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:58] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:58] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:58] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:58] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58b1e37a Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Contact: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] VERBOSE[13292] dial.c: SIP/zvonobot-0000003b answered [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Finding handler for bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/addChannel [Aug 18 10:33:58] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP creating BEGIN DTMF Frame: 55 (7), at 178.62.121.41:16938 [Aug 18 10:33:58] DTMF[13454] channel.c: DTMF begin '7' received on SIP/zvonobot-00000023 [Aug 18 10:33:58] DTMF[13454] channel.c: DTMF begin passthrough '7' on SIP/zvonobot-00000023 [Aug 18 10:33:58] DEBUG[13614] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] VERBOSE[13292] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003b [Aug 18 10:33:58] DEBUG[13610] http.c: match request [ari/channels/213045] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13292] stasis/app.c: Channel '213023' is 2 interested in calls_0 [Aug 18 10:33:58] DEBUG[13618] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13594] stasis.c: Creating topic. name: channel:213048, detail: [Aug 18 10:33:58] DEBUG[13594] stasis.c: Topic 'channel:213048': 0x7f0cb404b1e0 created [Aug 18 10:33:58] DEBUG[13594] stasis.c: Creating topic. name: cache:189/channel:213048, detail: [Aug 18 10:33:58] DEBUG[13594] stasis.c: Topic 'cache:189/channel:213048': 0x7f0cb4065270 created [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Finding handler for robot_212993 [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking channels create: Didn't match robot_212993 [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13608] res_ari.c: Checking channels externalMedia: Didn't match robot_212993 [Aug 18 10:33:58] DEBUG[13608] res_ari.c: No explicit handler found for robot_212993. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Finding handler for e594e1d1-53fe-4904-8517-472d8e3b8b52 [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13609] res_ari.c: No explicit handler found for e594e1d1-53fe-4904-8517-472d8e3b8b52. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Finding handler for addChannel [Aug 18 10:33:58] DEBUG[13609] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:58] DEBUG[13609] stasis/control.c: 213011: Sending channel add_to_bridge command [Aug 18 10:33:58] DEBUG[13606] rtp_engine.c: RTP instance '0x7f0c100f9ed0' is setup and ready to go [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE stopped [Aug 18 10:33:58] VERBOSE[13232] res_rtp_asterisk.c: 0x7f0c3401d7e0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:18326 [Aug 18 10:33:58] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTP creating BEGIN DTMF Frame: 54 (6), at 178.62.121.41:18326 [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF begin '6' received on SIP/zvonobot-00000016 [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF begin passthrough '6' on SIP/zvonobot-00000016 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Finding handler for 213051 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking channels create: Didn't match 213051 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13602] res_ari.c: Checking channels externalMedia: Didn't match 213051 [Aug 18 10:33:58] DEBUG[13602] res_ari.c: No explicit handler found for 213051. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13618] http.c: HTTP Request URI is /ari/channels/213050?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116990&callerId=74950493843 [Aug 18 10:33:58] DEBUG[13616] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13610] http.c: match request [ari/channels/213045] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13618] http.c: match request [ari/channels/213050] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13606] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13613] http.c: match request [ari/channels/213052] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13613] http.c: match request [ari/channels/213052] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13610] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13618] http.c: match request [ari/channels/213050] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13613] http.c: match request [ari/channels/213052] with handler [ari] len 3 [Aug 18 10:33:58] VERBOSE[13292] res_rtp_asterisk.c: 0x7f0c7c037ee0 -- Strict RTP switching to RTP target address 178.62.121.41:10224 as source [Aug 18 10:33:58] DEBUG[13292] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:58] DEBUG[13292] channel.c: Channel SIP/zvonobot-0000003b setting read format path: ulaw -> alaw [Aug 18 10:33:58] DEBUG[13292] channel.c: Channel SIP/zvonobot-0000003b setting write format path: alaw -> ulaw [Aug 18 10:33:58] DEBUG[13201] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000030 [Aug 18 10:33:58] DEBUG[13201] stasis/control.c: 213011: Adding to bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 [Aug 18 10:33:58] DEBUG[13201] stasis/app.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[13619] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is joining [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:58] DEBUG[13499] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13606] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13606] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13606] res_rtp_asterisk.c: (0x7f0c100f9ed0) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13606] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13606] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: SIP call-id changed from '20a251b0721862b6130f69bd013031b8@127.0.1.1:5060' to '59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13606] stasis.c: Creating topic. name: channel:213047, detail: [Aug 18 10:33:58] DEBUG[13606] stasis.c: Topic 'channel:213047': 0x7f0c10108470 created [Aug 18 10:33:58] DEBUG[13606] stasis.c: Creating topic. name: cache:190/channel:213047, detail: [Aug 18 10:33:58] DEBUG[13606] stasis.c: Topic 'cache:190/channel:213047': 0x7f0c10108ef0 created [Aug 18 10:33:58] DEBUG[13616] http.c: HTTP Request URI is /ari/channels/213049?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116991&callerId=74950493843 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:58] DEBUG[13613] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13610] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13616] http.c: match request [ari/channels/213049] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DTMF[13465] channel.c: DTMF begin '7' received on Recorder/ARI-00000013;1 [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DTMF[13465] channel.c: DTMF begin passthrough '7' on Recorder/ARI-00000013;1 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033 Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117005@178.62.121.41", nonce="17c0a9ea", response="7c1cb2cc0cbfa4cd0b231dd890a72a4f" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13618] http.c: match request [ari/channels/213050] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Finding handler for channels/213045 [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] http.c: match request [ari/channels/213049] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13619] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pushing 0x7f0ca0073e00(SIP/zvonobot-00000030) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[13613] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Finding handler for channels/213053 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DTMF[13247] channel.c: DTMF begin '6' received on Announcer/ARI-00000006;1 [Aug 18 10:33:58] DTMF[13247] channel.c: DTMF begin ignored '6' on Announcer/ARI-00000006;1 [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Finding handler for channels/213052 [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Finding handler for 213052 [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking channels create: Didn't match 213052 [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13613] res_ari.c: Checking channels externalMedia: Didn't match 213052 [Aug 18 10:33:58] DEBUG[13613] res_ari.c: No explicit handler found for 213052. Using wildcard channelId. [Aug 18 10:33:58] VERBOSE[13347] res_rtp_asterisk.c: 0x7f0c4000d2b0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:16138 [Aug 18 10:33:58] DTMF[13235] channel.c: DTMF begin '6' received on Recorder/ARI-00000005;1 [Aug 18 10:33:58] DTMF[13235] channel.c: DTMF begin passthrough '6' on Recorder/ARI-00000005;1 [Aug 18 10:33:58] DEBUG[13611] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13611] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13616] http.c: match request [ari/channels/213049] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117002@178.62.121.41", nonce="3423c0a6", response="6b620ac742be9e83a3506227f8751ecd" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416057 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[13620] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13611] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play] with handler [ari] len 3 [Aug 18 10:33:58] VERBOSE[13619] bridge_channel.c: Channel SIP/zvonobot-00000030 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTP creating END DTMF Frame: 54 (6), at 178.62.121.41:18326 [Aug 18 10:33:58] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 494 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 494 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF end '6' received on SIP/zvonobot-00000016, duration 140 ms [Aug 18 10:33:58] DEBUG[13611] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13618] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13618] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Session timer started: 39 - 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 1768000ms [Aug 18 10:33:58] DEBUG[13616] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Finding handler for 213053 [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking channels create: Didn't match 213053 [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13614] res_ari.c: Checking channels externalMedia: Didn't match 213053 [Aug 18 10:33:58] DEBUG[13614] res_ari.c: No explicit handler found for 213053. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13620] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF end accepted with begin '6' on SIP/zvonobot-00000016 [Aug 18 10:33:58] DEBUG[13620] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DTMF[13532] channel.c: DTMF begin '7' received on Announcer/ARI-00000017;1 [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13620] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13620] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Finding handler for channels/213050 [Aug 18 10:33:58] DTMF[13532] channel.c: DTMF begin ignored '7' on Announcer/ARI-00000017;1 [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF end '6' detected to have actual duration 59 on the wire, emulation will be triggered on SIP/zvonobot-00000016 [Aug 18 10:33:58] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 688, ms is 63 [Aug 18 10:33:58] DEBUG[13616] http.c: HTTP consuming request body [Aug 18 10:33:58] DEBUG[13620] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13621] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Finding handler for 213045 [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Finding handler for channels [Aug 18 10:33:58] VERBOSE[13349] res_rtp_asterisk.c: 0x7f0c70015c10 -- Strict RTP learning complete - Locking on source address 178.62.121.41:19990 [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking channels create: Didn't match 213045 [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13610] res_ari.c: Checking channels externalMedia: Didn't match 213045 [Aug 18 10:33:58] DEBUG[13610] res_ari.c: No explicit handler found for 213045. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 From: ;tag=as2b432b30 To: ;tag=as069541b6 Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0346307c" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[13507] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF end '6' has duration 59 but want minimum 80, emulating on SIP/zvonobot-00000016 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as069541b6 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0346307c" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag as069541b6 [Aug 18 10:33:58] DTMF[13232] channel.c: DTMF end emulation of '6' queued on SIP/zvonobot-00000016 [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Finding handler for bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162/play [Aug 18 10:33:58] DEBUG[13621] http.c: HTTP Request URI is /ari/playbacks/b1e337c9-b870-44c1-95a4-86716e990798 [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP creating END DTMF Frame: 55 (7), at 178.62.121.41:16938 [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Finding handler for channels/213049 [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be Max-Forwards: 70 From: ;tag=as2b432b30 To: ;tag=as069541b6 Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Audio is at 10382 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 494 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 494 [Aug 18 10:33:58] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13620] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117001@178.62.121.41", nonce="0346307c", response="cf82a81f12417d8ec0d755f08d1960a5" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437934 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #37 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #47 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #47)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160452 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #41 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #41)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010121 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437933 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16;received=159.65.48.104 From: ;tag=as3edf3f1c To: ;tag=as177609bc Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e114138" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as177609bc [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e114138" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag as177609bc [Aug 18 10:33:58] DEBUG[13314] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13620] stasis.c: Creating topic. name: bridge:357a4882-a24d-489f-8ff8-98badd81b2ee, detail: [Aug 18 10:33:58] DEBUG[13620] stasis.c: Topic 'bridge:357a4882-a24d-489f-8ff8-98badd81b2ee': 0x7f0c3c060df0 created [Aug 18 10:33:58] DEBUG[13620] stasis.c: Creating topic. name: cache:191/bridge:357a4882-a24d-489f-8ff8-98badd81b2ee, detail: [Aug 18 10:33:58] DEBUG[13620] stasis.c: Topic 'cache:191/bridge:357a4882-a24d-489f-8ff8-98badd81b2ee': 0x7f0c3c043610 created [Aug 18 10:33:58] DEBUG[13620] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13620] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13620] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13620] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:58] DEBUG[13620] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13620] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology constructor [Aug 18 10:33:58] DEBUG[13620] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology start [Aug 18 10:33:58] DTMF[13247] channel.c: DTMF end '6' received on Announcer/ARI-00000006;1, duration 114 ms [Aug 18 10:33:58] DTMF[13247] channel.c: DTMF end passthrough '6' on Announcer/ARI-00000006;1 [Aug 18 10:33:58] DTMF[13454] channel.c: DTMF end '7' received on SIP/zvonobot-00000023, duration 140 ms [Aug 18 10:33:58] DTMF[13454] channel.c: DTMF end accepted with begin '7' on SIP/zvonobot-00000023 [Aug 18 10:33:58] DTMF[13454] channel.c: DTMF end passthrough '7' on SIP/zvonobot-00000023 [Aug 18 10:33:58] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #48 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16 Max-Forwards: 70 From: ;tag=as3edf3f1c To: ;tag=as177609bc Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Audio is at 15562 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117006@178.62.121.41", nonce="2e114138", response="546d7b2d9fa7f2afdf06a7c01e9051cf" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379123 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #33 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DTMF[13235] channel.c: DTMF end '6' received on Recorder/ARI-00000005;1, duration 114 ms [Aug 18 10:33:58] DTMF[13235] channel.c: DTMF end accepted with begin '6' on Recorder/ARI-00000005;1 [Aug 18 10:33:58] DTMF[13235] channel.c: DTMF end passthrough '6' on Recorder/ARI-00000005;1 [Aug 18 10:33:58] DEBUG[13623] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13623] http.c: HTTP Request URI is /ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/addChannel?channel=213023 [Aug 18 10:33:58] DEBUG[13623] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13623] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: Allocating new SIP dialog for 2abe385f3ffac9282dfdc29d27f226c4@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13623] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/addChannel] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13431] channel.c: Channel 0x7f0c78050c90 'Announcer/ARI-00000010;2' destroying [Aug 18 10:33:58] DEBUG[13431] stasis.c: Destroying topic. name: cache:142/channel:1629282834.121, detail: [Aug 18 10:33:58] DEBUG[13431] stasis.c: Topic 'cache:142/channel:1629282834.121': 0x7f0c78019cb0 destroyed [Aug 18 10:33:58] DEBUG[13431] stasis.c: Destroying topic. name: channel:1629282834.121, detail: [Aug 18 10:33:58] DEBUG[13431] stasis.c: Topic 'channel:1629282834.121': 0x7f0c78019b00 destroyed [Aug 18 10:33:58] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13044] chan_sip.c: Hangup call SIP/zvonobot-0000001b, SIP callid 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13044] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[13044] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[13558] channel.c: Channel 0x7f0c7c077520 'Recorder/ARI-00000019;2' allocated [Aug 18 10:33:58] DEBUG[13558] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:58] DEBUG[13562] channel.c: Channel 0x7f0c80047da0 'Recorder/ARI-0000001a;2' allocated [Aug 18 10:33:58] DEBUG[13562] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:58] DEBUG[13626] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining [Aug 18 10:33:58] DEBUG[13602] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c67ec0' [Aug 18 10:33:58] DEBUG[13621] http.c: match request [ari/playbacks/b1e337c9-b870-44c1-95a4-86716e990798] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: Allocating new SIP dialog for 1e33050427a7a3de5e9933267f766975@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13614] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c0862c0' [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) RTP allocated port 14624 [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE creating session 0.0.0.0:14624 (14624) [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE create [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE add system candidates [Aug 18 10:33:58] DEBUG[13614] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13614] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE add candidate: 159.65.48.104:14624, 2130706431 [Aug 18 10:33:58] DEBUG[13614] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13614] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE add candidate: 10.131.0.10:14624, 2130706431 [Aug 18 10:33:58] DEBUG[13614] rtp_engine.c: RTP instance '0x7f0c2c0862c0' is setup and ready to go [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE stopped [Aug 18 10:33:58] DEBUG[13614] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13614] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13614] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13614] res_rtp_asterisk.c: (0x7f0c2c0862c0) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13614] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13614] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13620] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DTMF[13465] channel.c: DTMF end '7' received on Recorder/ARI-00000013;1, duration 140 ms [Aug 18 10:33:58] DTMF[13465] channel.c: DTMF end accepted with begin '7' on Recorder/ARI-00000013;1 [Aug 18 10:33:58] DTMF[13465] channel.c: DTMF end passthrough '7' on Recorder/ARI-00000013;1 [Aug 18 10:33:58] DEBUG[13314] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13314] bridge_channel.c: Setting 0x7f0cb4036220(Snoop/212993-00000003) state from:0 to:1 [Aug 18 10:33:58] DEBUG[13620] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) RTP allocated port 15904 [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE creating session 0.0.0.0:15904 (15904) [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE create [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE add system candidates [Aug 18 10:33:58] DEBUG[13602] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13602] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE add candidate: 159.65.48.104:15904, 2130706431 [Aug 18 10:33:58] DEBUG[13602] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13602] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE add candidate: 10.131.0.10:15904, 2130706431 [Aug 18 10:33:58] DEBUG[13602] rtp_engine.c: RTP instance '0x2c67ec0' is setup and ready to go [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) ICE stopped [Aug 18 10:33:58] DEBUG[13602] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13602] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13602] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13602] res_rtp_asterisk.c: (0x2c67ec0) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13602] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13602] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Finding handler for 213049 [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking channels create: Didn't match 213049 [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13616] res_ari.c: Checking channels externalMedia: Didn't match 213049 [Aug 18 10:33:58] DEBUG[13616] res_ari.c: No explicit handler found for 213049. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423 Max-Forwards: 70 From: ;tag=as46ab0e55 To: ;tag=as396a139d Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="39509751", response="ab8220e1f7ce5805733f6c6a4eac6fb4" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[13624] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining [Aug 18 10:33:58] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13507] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13557] stasis.c: Creating topic. name: channel:1629282838.161, detail: [Aug 18 10:33:58] DEBUG[13557] stasis.c: Topic 'channel:1629282838.161': 0x7f0c70070650 created [Aug 18 10:33:58] DEBUG[13557] stasis.c: Creating topic. name: cache:192/channel:1629282838.161, detail: [Aug 18 10:33:58] DEBUG[13557] stasis.c: Topic 'cache:192/channel:1629282838.161': 0x7f0c70073a00 created [Aug 18 10:33:58] DEBUG[13621] http.c: match request [ari/playbacks/b1e337c9-b870-44c1-95a4-86716e990798] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13623] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13621] http.c: match request [ari/playbacks/b1e337c9-b870-44c1-95a4-86716e990798] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13455] channel.c: Channel 0x7f0c300a36a0 'Announcer/ARI-00000012;2' destroying [Aug 18 10:33:58] DEBUG[13621] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 640, ms is 60 [Aug 18 10:33:58] DEBUG[13314] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: pulling 0x7f0cb4036220(Snoop/212993-00000003) [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:58] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:33:58] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:33:58] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:58] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 816, ms is 71 [Aug 18 10:33:58] DTMF[13532] channel.c: DTMF end '7' received on Announcer/ARI-00000017;1, duration 140 ms [Aug 18 10:33:58] DTMF[13532] channel.c: DTMF end passthrough '7' on Announcer/ARI-00000017;1 [Aug 18 10:33:58] DEBUG[13433] channel.c: Channel 0x7f0c780058a0 'Announcer/ARI-00000010;1' destroying [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:33:58] DEBUG[13619] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13619] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13619] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13619] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13619] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13619] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 is already using the new technology. [Aug 18 10:33:58] DEBUG[13619] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Finding handler for bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/addChannel [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Finding handler for 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13623] res_ari.c: No explicit handler found for 357a4882-a24d-489f-8ff8-98badd81b2ee. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Finding handler for addChannel [Aug 18 10:33:58] DEBUG[13623] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:58] DEBUG[13623] stasis/control.c: 213023: Sending channel add_to_bridge command [Aug 18 10:33:58] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13292] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000003b [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: SIP call-id changed from '2abe385f3ffac9282dfdc29d27f226c4@127.0.1.1:5060' to '16541045053beef31ed8e5361337aa22@159.65.48.104:5060' [Aug 18 10:33:58] VERBOSE[13314] bridge_channel.c: Channel Snoop/212993-00000003 left 'simple_bridge' stasis-bridge <9d1bf1e2-893f-4249-b006-4b3a345e76a2> [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: SIP call-id changed from '1e33050427a7a3de5e9933267f766975@127.0.1.1:5060' to '63c7430f0c6f8039194559375e330124@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13614] stasis.c: Creating topic. name: channel:213053, detail: [Aug 18 10:33:58] DEBUG[13614] stasis.c: Topic 'channel:213053': 0x7f0c2c0930c0 created [Aug 18 10:33:58] DEBUG[13614] stasis.c: Creating topic. name: cache:193/channel:213053, detail: [Aug 18 10:33:58] DEBUG[13614] stasis.c: Topic 'cache:193/channel:213053': 0x7f0c2c093ab0 created [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: Allocating new SIP dialog for 0fee7ed866536e23695576ca4870fc6e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13589] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb00e8f80' [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) RTP allocated port 10612 [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE creating session 0.0.0.0:10612 (10612) [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE create [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE add system candidates [Aug 18 10:33:58] DEBUG[13589] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13589] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13541] res_rtp_asterisk.c: (0x7f0c080871b0) RTP audio difference is 736, ms is 66 [Aug 18 10:33:58] DEBUG[13602] stasis.c: Creating topic. name: channel:213051, detail: [Aug 18 10:33:58] DEBUG[13298] channel.c: Channel 0x7f0c980450a0 'Recorder/ARI-00000007;2' destroying [Aug 18 10:33:58] DEBUG[13433] stasis.c: Destroying topic. name: cache:141/channel:1629282834.120, detail: [Aug 18 10:33:58] DEBUG[13433] stasis.c: Topic 'cache:141/channel:1629282834.120': 0x7f0c7804b7c0 destroyed [Aug 18 10:33:58] VERBOSE[13299] app.c: User hung up [Aug 18 10:33:58] DEBUG[13314] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0cb4036220(Snoop/212993-00000003) is leaving simple_bridge technology [Aug 18 10:33:58] DEBUG[13247] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13292] stasis/control.c: 213023: Adding to bridge 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Finding handler for 213050 [Aug 18 10:33:58] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13299] res_stasis_recording.c: 1629282832.83: Recording complete [Aug 18 10:33:58] DEBUG[13235] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13499] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13602] stasis.c: Topic 'channel:213051': 0x2c44f10 created [Aug 18 10:33:58] DEBUG[13602] stasis.c: Creating topic. name: cache:194/channel:213051, detail: [Aug 18 10:33:58] DEBUG[13602] stasis.c: Topic 'cache:194/channel:213051': 0x2c3de20 created [Aug 18 10:33:58] DEBUG[13251] res_rtp_asterisk.c: (0x7f0ca0023720) RTP audio difference is 768, ms is 68 [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13454] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423 [Aug 18 10:33:58] DEBUG[13292] stasis/app.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Finding handler for playbacks/b1e337c9-b870-44c1-95a4-86716e990798 [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking channels create: Didn't match 213050 [Aug 18 10:33:58] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 960, ms is 80 [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: Allocating new SIP dialog for 7ba50575273ca0336d03e4c34e2bcad9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13610] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c180c9c20' [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) RTP allocated port 13804 [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE creating session 0.0.0.0:13804 (13804) [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE create [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE add system candidates [Aug 18 10:33:58] DEBUG[13610] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13610] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE add candidate: 159.65.48.104:13804, 2130706431 [Aug 18 10:33:58] DEBUG[13610] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13610] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE add candidate: 10.131.0.10:13804, 2130706431 [Aug 18 10:33:58] DEBUG[13610] rtp_engine.c: RTP instance '0x7f0c180c9c20' is setup and ready to go [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE stopped [Aug 18 10:33:58] DEBUG[13610] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13610] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 880, ms is 75 [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE add candidate: 159.65.48.104:10612, 2130706431 [Aug 18 10:33:58] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13589] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13589] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE add candidate: 10.131.0.10:10612, 2130706431 [Aug 18 10:33:58] DEBUG[13589] rtp_engine.c: RTP instance '0x7f0cb00e8f80' is setup and ready to go [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE stopped [Aug 18 10:33:58] DEBUG[13589] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13589] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13589] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13589] res_rtp_asterisk.c: (0x7f0cb00e8f80) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13589] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13299] channel.c: Channel 0x7f0c9803f2e0 'Recorder/ARI-00000007;1' hanging up. Refs: 2 [Aug 18 10:33:58] DEBUG[13504] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13618] res_ari.c: Checking channels externalMedia: Didn't match 213050 [Aug 18 10:33:58] DEBUG[13618] res_ari.c: No explicit handler found for 213050. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:58] DEBUG[13610] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13627] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is joining [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:58] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: Allocating new SIP dialog for 0c5ccf8b106efafe2459ecef097e7703@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13232] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP got report of 100 bytes from 178.62.121.41:12913 [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:58] DEBUG[13433] stasis.c: Destroying topic. name: channel:1629282834.120, detail: [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Finding handler for playbacks [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13433] stasis.c: Topic 'channel:1629282834.120': 0x7f0c7802c9a0 destroyed [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP changing ssrc from 870389521 to 560879787 due to a source change [Aug 18 10:33:58] DEBUG[13504] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13589] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13589] chan_sip.c: SIP call-id changed from '0fee7ed866536e23695576ca4870fc6e@127.0.1.1:5060' to '353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13589] stasis.c: Creating topic. name: channel:213044, detail: [Aug 18 10:33:58] DEBUG[13589] stasis.c: Topic 'channel:213044': 0x7f0cb00f60a0 created [Aug 18 10:33:58] DEBUG[13589] stasis.c: Creating topic. name: cache:195/channel:213044, detail: [Aug 18 10:33:58] DEBUG[13589] stasis.c: Topic 'cache:195/channel:213044': 0x7f0cb00e3b60 created [Aug 18 10:33:58] DEBUG[13201] stasis/app.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' is 2 interested in calls_0 [Aug 18 10:33:58] DEBUG[13624] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pushing 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13298] stasis.c: Destroying topic. name: cache:98/channel:1629282832.84, detail: [Aug 18 10:33:58] DEBUG[13298] stasis.c: Topic 'cache:98/channel:1629282832.84': 0x7f0c9803ccc0 destroyed [Aug 18 10:33:58] DEBUG[13298] stasis.c: Destroying topic. name: channel:1629282832.84, detail: [Aug 18 10:33:58] DEBUG[13298] stasis.c: Topic 'channel:1629282832.84': 0x7f0c9802db40 destroyed [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13626] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pushing 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as46ab0e55 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as396a139d [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="39509751", response="ab8220e1f7ce5805733f6c6a4eac6fb4" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 (Checking From) --From tag as46ab0e55 --To-tag as396a139d [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:58] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Received bye, no owner, selfdestruct soon. [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK67eb0423;received=178.62.121.41 From: ;tag=as46ab0e55 To: ;tag=as396a139d Call-ID: 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801089 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033 Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117005@178.62.121.41", nonce="17c0a9ea", response="7c1cb2cc0cbfa4cd0b231dd890a72a4f" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1899292571 1899292572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #41 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #41)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010121 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117001@178.62.121.41", nonce="0346307c", response="cf82a81f12417d8ec0d755f08d1960a5" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437934 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13613] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c20075860' [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) RTP allocated port 14750 [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE creating session 0.0.0.0:14750 (14750) [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE create [Aug 18 10:33:58] DEBUG[13610] res_rtp_asterisk.c: (0x7f0c180c9c20) RTCP setup on RTP instance [Aug 18 10:33:58] DEBUG[13628] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13609] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:58] VERBOSE[13610] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212993-00000003 - start 1629282832.445521 answer 1629282832.445521 end 1629282838.468950 dur 6.023 bill 6.023 dispo ANSWERED [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13314] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13314] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13314] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13314] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13314] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13314] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 is already using the new technology. [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: Allocating new SIP dialog for 56e84cd07bd2835928fa75bc2aea3e80@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13592] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac0660c0' [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) RTP allocated port 17196 [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE creating session 0.0.0.0:17196 (17196) [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE create [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE add system candidates [Aug 18 10:33:58] DEBUG[13592] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13592] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE add candidate: 159.65.48.104:17196, 2130706431 [Aug 18 10:33:58] DEBUG[13592] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13592] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE add candidate: 10.131.0.10:17196, 2130706431 [Aug 18 10:33:58] DEBUG[13592] rtp_engine.c: RTP instance '0x7f0cac0660c0' is setup and ready to go [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE stopped [Aug 18 10:33:58] DEBUG[13592] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13592] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13592] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13592] res_rtp_asterisk.c: (0x7f0cac0660c0) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13592] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13592] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13592] chan_sip.c: SIP call-id changed from '56e84cd07bd2835928fa75bc2aea3e80@127.0.1.1:5060' to '68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13592] stasis.c: Creating topic. name: channel:213046, detail: [Aug 18 10:33:58] DEBUG[13592] stasis.c: Topic 'channel:213046': 0x7f0cac07bcd0 created [Aug 18 10:33:58] DEBUG[13592] stasis.c: Creating topic. name: cache:196/channel:213046, detail: [Aug 18 10:33:58] DEBUG[13592] stasis.c: Topic 'cache:196/channel:213046': 0x7f0cac07c750 created [Aug 18 10:33:58] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13609] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117006@178.62.121.41", nonce="2e114138", response="546d7b2d9fa7f2afdf06a7c01e9051cf" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379123 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 From: ;tag=as22c76af6 To: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244;received=159.65.48.104 From: ;tag=as6ceaa437 To: ;tag=as17140454 Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="048e446a" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as17140454 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="048e446a" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag as17140454 [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE add system candidates [Aug 18 10:33:58] DEBUG[13613] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13613] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE add candidate: 159.65.48.104:14750, 2130706431 [Aug 18 10:33:58] DEBUG[13613] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13613] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE add candidate: 10.131.0.10:14750, 2130706431 [Aug 18 10:33:58] DEBUG[13613] rtp_engine.c: RTP instance '0x7f0c20075860' is setup and ready to go [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) ICE stopped [Aug 18 10:33:58] DEBUG[13613] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13613] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13613] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13613] res_rtp_asterisk.c: (0x7f0c20075860) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13613] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13613] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13613] chan_sip.c: SIP call-id changed from '0c5ccf8b106efafe2459ecef097e7703@127.0.1.1:5060' to '10507dcf059680b46ad884550335c862@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13613] stasis.c: Creating topic. name: channel:213052, detail: [Aug 18 10:33:58] DEBUG[13613] stasis.c: Topic 'channel:213052': 0x7f0c200836d0 created [Aug 18 10:33:58] DEBUG[13613] stasis.c: Creating topic. name: cache:197/channel:213052, detail: [Aug 18 10:33:58] DEBUG[13613] stasis.c: Topic 'cache:197/channel:213052': 0x7f0c200840d0 created [Aug 18 10:33:58] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 1056, ms is 86 [Aug 18 10:33:58] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Finding handler for b1e337c9-b870-44c1-95a4-86716e990798 [Aug 18 10:33:58] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13232] audiohook.c: Audiohook 0x7f0ca802f950 has stale audio in its factories. Flushing them both [Aug 18 10:33:58] DEBUG[13621] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244 Max-Forwards: 70 From: ;tag=as6ceaa437 To: ;tag=as17140454 Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Audio is at 11300 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116999@178.62.121.41", nonce="048e446a", response="229b027e812dcf7d91509c351fa56509" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa;received=159.65.48.104 From: ;tag=as30c9fc1b To: ;tag=as1917d2c9 Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17c0a9ea" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20142bfa;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1917d2c9 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17c0a9ea" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag as1917d2c9 [Aug 18 10:33:58] DEBUG[13610] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Finding handler for 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13611] res_ari.c: No explicit handler found for 3d1f9573-48d1-48a0-bcdd-9d7f09555162. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Finding handler for play [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:58] DEBUG[13611] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:58] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117005@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033 Max-Forwards: 70 From: ;tag=as30c9fc1b To: Contact: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13628] http.c: HTTP Request URI is /ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/record?name=213011_NLuXdhmeVPfdDxEObkwOLqvNJGIULStr&format=wav [Aug 18 10:33:58] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13621] res_ari.c: No explicit handler found for b1e337c9-b870-44c1-95a4-86716e990798. Using wildcard playbackId. [Aug 18 10:33:58] DEBUG[13621] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: Allocating new SIP dialog for 2cf3b29c16bd7f5139f829744c935a71@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13616] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280ef5e0' [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:33:58] DEBUG[13621] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13247] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13247] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13314] bridge_channel.c: Bridge is returning 0x7f0cb4036220(Snoop/212993-00000003) to read format slin [Aug 18 10:33:58] DEBUG[13455] stasis.c: Destroying topic. name: cache:151/channel:1629282835.126, detail: [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) RTP allocated port 15836 [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE creating session 0.0.0.0:15836 (15836) [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE create [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 From: ;tag=as30c9fc1b To: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #42 - INVITE (got response) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13455] stasis.c: Topic 'cache:151/channel:1629282835.126': 0x7f0c30071ce0 destroyed [Aug 18 10:33:58] DEBUG[13314] channel.c: Channel Snoop/212993-00000003 setting read format path: slin -> slin [Aug 18 10:33:58] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE add system candidates [Aug 18 10:33:58] DEBUG[13616] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13616] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE add candidate: 159.65.48.104:15836, 2130706431 [Aug 18 10:33:58] DEBUG[13616] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13314] bridge_channel.c: Bridge is returning 0x7f0cb4036220(Snoop/212993-00000003) to write format slin [Aug 18 10:33:58] DEBUG[13630] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13247] channel.c: Channel Announcer/ARI-00000006;1 setting write format path: slin -> slin [Aug 18 10:33:58] NOTICE[13247] res_stasis_playback.c: 1629282831.70: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:33:58] DEBUG[13247] channel.c: Channel 0x7f0c9802bfd0 'Announcer/ARI-00000006;1' hanging up. Refs: 2 [Aug 18 10:33:58] DEBUG[13455] stasis.c: Destroying topic. name: channel:1629282835.126, detail: [Aug 18 10:33:58] DEBUG[13616] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13628] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/record] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:33:58] DEBUG[13314] channel.c: Channel Snoop/212993-00000003 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 688, ms is 106 [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: Allocating new SIP dialog for 085ae8004505dbc40b5c08f31c4a969d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:33:58] DEBUG[13618] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c340ab160' [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) RTP allocated port 10086 [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE creating session 0.0.0.0:10086 (10086) [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE create [Aug 18 10:33:58] DEBUG[13455] stasis.c: Topic 'channel:1629282835.126': 0x7f0c3007c7b0 destroyed [Aug 18 10:33:58] DEBUG[13314] stasis/control.c: 1629282832.85, 9d1bf1e2-893f-4249-b006-4b3a345e76a2: Channel was departed from bridge [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE add candidate: 10.131.0.10:15836, 2130706431 [Aug 18 10:33:58] DEBUG[13616] rtp_engine.c: RTP instance '0x7f0c280ef5e0' is setup and ready to go [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE stopped [Aug 18 10:33:58] DEBUG[13616] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13616] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13616] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13616] res_rtp_asterisk.c: (0x7f0c280ef5e0) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13616] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13616] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13628] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/record] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13627] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pushing 0x7f0ca803dbf0(SIP/zvonobot-0000003b) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 From: ;tag=as22c76af6 To: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag [Aug 18 10:33:58] DEBUG[13630] http.c: HTTP Request URI is /ari/channels/robot_212986 [Aug 18 10:33:58] DEBUG[13314] stasis/app.c: bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2': is 3 interested in calls_0 [Aug 18 10:33:58] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13628] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/record] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13624] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:58] VERBOSE[13624] bridge_channel.c: Channel Recorder/ARI-00000019;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13626] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:58] VERBOSE[13626] bridge_channel.c: Channel Recorder/ARI-0000001a;2 joined 'simple_bridge' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:33:58] DEBUG[13630] http.c: match request [ari/channels/robot_212986] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13628] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13302] stasis/control.c: 1629282832.85: Channel departing bridge [Aug 18 10:33:58] DEBUG[13302] bridge.c: Waiting for 0x7f0cb4036220(Snoop/212993-00000003) bridge thread to die. [Aug 18 10:33:58] DEBUG[13314] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Finding handler for bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/record [Aug 18 10:33:58] DEBUG[13630] http.c: match request [ari/channels/robot_212986] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13302] stasis/app.c: channel '1629282832.85': is 0 interested in calls_0 [Aug 18 10:33:58] DEBUG[13302] stasis/app.c: channel '1629282832.85' unsubscribed from calls_0 [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE add system candidates [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 445 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 445 [Aug 18 10:33:58] DEBUG[13618] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:58] DEBUG[13618] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE add candidate: 159.65.48.104:10086, 2130706431 [Aug 18 10:33:58] DEBUG[13618] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:58] DEBUG[13618] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Finding handler for e594e1d1-53fe-4904-8517-472d8e3b8b52 [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13628] res_ari.c: No explicit handler found for e594e1d1-53fe-4904-8517-472d8e3b8b52. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Finding handler for record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:58] DEBUG[13628] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:58] DEBUG[13628] stasis.c: Creating topic. name: channel:1629282838.167, detail: [Aug 18 10:33:58] DEBUG[13610] chan_sip.c: SIP call-id changed from '7ba50575273ca0336d03e4c34e2bcad9@127.0.1.1:5060' to '54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13610] stasis.c: Creating topic. name: channel:213045, detail: [Aug 18 10:33:58] DEBUG[13610] stasis.c: Topic 'channel:213045': 0x7f0c180c4730 created [Aug 18 10:33:58] DEBUG[13610] stasis.c: Creating topic. name: cache:198/channel:213045, detail: [Aug 18 10:33:58] DEBUG[13610] stasis.c: Topic 'cache:198/channel:213045': 0x7f0c1808f6e0 created [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:58] DEBUG[13302] channel.c: Channel 0x7f0cac03e430 'Snoop/212993-00000003' hanging up. Refs: 3 [Aug 18 10:33:58] DEBUG[13630] http.c: match request [ari/channels/robot_212986] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13628] stasis.c: Topic 'channel:1629282838.167': 0x7f0c78074b20 created [Aug 18 10:33:58] DEBUG[13628] stasis.c: Creating topic. name: cache:199/channel:1629282838.167, detail: [Aug 18 10:33:58] DEBUG[13628] stasis.c: Topic 'cache:199/channel:1629282838.167': 0x7f0c78075540 created [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3;received=159.65.48.104 From: ;tag=as19ef62c2 To: ;tag=as062fa867 Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3423c0a6" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45de76c3;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as062fa867 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3423c0a6" [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag as062fa867 [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13630] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13616] chan_sip.c: SIP call-id changed from '2cf3b29c16bd7f5139f829744c935a71@127.0.1.1:5060' to '50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE add candidate: 10.131.0.10:10086, 2130706431 [Aug 18 10:33:58] VERBOSE[13627] bridge_channel.c: Channel SIP/zvonobot-0000003b joined 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:33:58] DEBUG[13618] rtp_engine.c: RTP instance '0x7f0c340ab160' is setup and ready to go [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Finding handler for channels/robot_212986 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117002@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652 Max-Forwards: 70 From: ;tag=as19ef62c2 To: Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117002@178.62.121.41", nonce="3423c0a6", response="6b620ac742be9e83a3506227f8751ecd" Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 258416056 258416057 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7;received=159.65.48.104 From: ;tag=as0e0b214d To: ;tag=as285b992f Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dbc5b08" Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as285b992f [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13624] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6'. Checking compatability for channels 'SIP/zvonobot-0000000e' and 'Recorder/ARI-00000019;2' [Aug 18 10:33:58] DEBUG[13624] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as could not get details [Aug 18 10:33:58] DEBUG[13624] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13624] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13624] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13624] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13624] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 is already using the new technology. [Aug 18 10:33:58] DEBUG[13624] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel Recorder/ARI-00000019;2 setting read format path: slin -> slin [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel SIP/zvonobot-0000000e setting write format path: slin -> ulaw [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel SIP/zvonobot-0000000e setting read format path: ulaw -> slin [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13501] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13616] stasis.c: Creating topic. name: channel:213049, detail: [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) ICE stopped [Aug 18 10:33:58] DEBUG[13618] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:33:58] DEBUG[13618] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:33:58] DEBUG[13618] acl.c: Multiple addresses. Using the first only [Aug 18 10:33:58] DEBUG[13618] res_rtp_asterisk.c: (0x7f0c340ab160) RTCP setup on RTP instance [Aug 18 10:33:58] VERBOSE[13618] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:33:58] DEBUG[13618] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:33:58] DEBUG[13618] chan_sip.c: SIP call-id changed from '085ae8004505dbc40b5c08f31c4a969d@127.0.1.1:5060' to '7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060' [Aug 18 10:33:58] DEBUG[13618] stasis.c: Creating topic. name: channel:213050, detail: [Aug 18 10:33:58] DEBUG[13626] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060'. Checking compatability for channels 'SIP/zvonobot-0000002b' and 'Recorder/ARI-0000001a;2' [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] DEBUG[13616] stasis.c: Topic 'channel:213049': 0x7f0c28106f70 created [Aug 18 10:33:58] DEBUG[13616] stasis.c: Creating topic. name: cache:200/channel:213049, detail: [Aug 18 10:33:58] DEBUG[13616] stasis.c: Topic 'cache:200/channel:213049': 0x7f0c281079f0 created [Aug 18 10:33:58] DEBUG[13618] stasis.c: Topic 'channel:213050': 0x7f0c340be5b0 created [Aug 18 10:33:58] DEBUG[13618] stasis.c: Creating topic. name: cache:201/channel:213050, detail: [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13594] channel.c: Channel 0x7f0cb40628c0 'SIP/zvonobot-00000050' allocated [Aug 18 10:33:58] DEBUG[13626] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as could not get details [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[13618] stasis.c: Topic 'cache:201/channel:213050': 0x7f0c340bf030 created [Aug 18 10:33:58] DEBUG[13626] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:58] DEBUG[13501] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dbc5b08" [Aug 18 10:33:58] DEBUG[13501] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13501] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:58] DEBUG[13501] channel.c: Channel Announcer/ARI-00000015;1 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13626] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13594] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag as285b992f [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13626] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Finding handler for robot_212986 [Aug 18 10:33:58] DEBUG[13626] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13626] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 is already using the new technology. [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking channels create: Didn't match robot_212986 [Aug 18 10:33:58] DEBUG[13627] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13626] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel Recorder/ARI-0000001a;2 setting read format path: slin -> slin [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel SIP/zvonobot-0000002b setting write format path: slin -> ulaw [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:58] DEBUG[13630] res_ari.c: Checking channels externalMedia: Didn't match robot_212986 [Aug 18 10:33:58] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTCP got report of 100 bytes from 178.62.121.41:16139 [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel SIP/zvonobot-0000002b setting read format path: ulaw -> slin [Aug 18 10:33:58] DEBUG[13630] res_ari.c: No explicit handler found for robot_212986. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [Aug 18 10:33:58] DEBUG[13501] channel.c: Channel 0x7f0c10068200 'Announcer/ARI-00000015;1' hanging up. Refs: 2 [Aug 18 10:33:58] DEBUG[13627] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:33:58] DEBUG[13627] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13606] channel.c: Channel 0x7f0c10106720 'SIP/zvonobot-00000051' allocated [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:58] DEBUG[13606] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7 Max-Forwards: 70 From: ;tag=as0e0b214d To: ;tag=as285b992f Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Audio is at 15196 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116997@178.62.121.41", nonce="2dbc5b08", response="c92db07f4aae82ff2863950559637b69" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485184 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #48 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[13627] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116999@178.62.121.41", nonce="048e446a", response="229b027e812dcf7d91509c351fa56509" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #47 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #47)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160452 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117001@178.62.121.41", nonce="0346307c", response="cf82a81f12417d8ec0d755f08d1960a5" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437934 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117006@178.62.121.41", nonce="2e114138", response="546d7b2d9fa7f2afdf06a7c01e9051cf" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379123 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13627] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13627] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee is already using the new technology. [Aug 18 10:33:58] DEBUG[13594] res_stasis.c: calls_0: Subscribing to 213048 [Aug 18 10:33:58] DEBUG[13627] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is joining simple_bridge technology [Aug 18 10:33:58] DEBUG[13594] stasis/app.c: Channel '213048' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[13594] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Outgoing Call for 79821116992 [Aug 18 10:33:58] DEBUG[13594] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13558] res_stasis_recording.c: 1629282837.153: Sending record(212977_vtgwVPwmpVRoqoJmilBCHrdNgFSFbIMS.wav) command [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:58] DEBUG[13558] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:58] DEBUG[13558] http.c: HTTP closing session. Top level [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 895903859 895903859 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10224 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ac5032e;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4f7b8e6e [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as121e4580 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[13634] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13584] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[13584] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP changing ssrc from 89419783 to 1668643816 due to a source change [Aug 18 10:33:58] DEBUG[13623] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:58] DEBUG[13623] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[13292] stasis/app.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' is 2 interested in calls_0 [Aug 18 10:33:58] DEBUG[13634] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:58] DEBUG[13634] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13634] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13634] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13634] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13634] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13634] stasis.c: Creating topic. name: bridge:48086187-3f40-424c-b978-0d6c6da7141b, detail: [Aug 18 10:33:58] DEBUG[13634] stasis.c: Topic 'bridge:48086187-3f40-424c-b978-0d6c6da7141b': 0x7f0c88072330 created [Aug 18 10:33:58] DEBUG[13634] stasis.c: Creating topic. name: cache:202/bridge:48086187-3f40-424c-b978-0d6c6da7141b, detail: [Aug 18 10:33:58] DEBUG[13634] stasis.c: Topic 'cache:202/bridge:48086187-3f40-424c-b978-0d6c6da7141b': 0x7f0c88072d30 created [Aug 18 10:33:58] DEBUG[13634] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13634] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13634] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13634] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:58] DEBUG[13634] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13632] app.c: play_and_record: , /var/spool/asterisk/recording/212977_vtgwVPwmpVRoqoJmilBCHrdNgFSFbIMS, 'wav' [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:58] DEBUG[13634] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: calling simple_bridge technology constructor [Aug 18 10:33:58] DEBUG[13635] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel Recorder/ARI-00000019;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] DEBUG[13632] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:58] VERBOSE[13631] chan_sip.c: Audio is at 18824 [Aug 18 10:33:58] DEBUG[13635] http.c: HTTP Request URI is /ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/record?name=213023_xsTqdZZuLTcscXpKpHHMfkbIildpHaqk&format=wav [Aug 18 10:33:58] DEBUG[13606] res_stasis.c: calls_0: Subscribing to 213047 [Aug 18 10:33:58] DEBUG[13606] stasis/app.c: Channel '213047' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:58] DEBUG[13606] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Outgoing Call for 79821116993 [Aug 18 10:33:58] DEBUG[13606] http.c: HTTP closing session. Top level [Aug 18 10:33:58] VERBOSE[13631] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] DEBUG[13634] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: calling simple_bridge technology start [Aug 18 10:33:58] DEBUG[13624] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: alaw -> slin [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 895903859 895903859 IN IP4 178.62.121.41 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10224 RTP/AVP 0 8 101 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 (Checking To) --From tag as4f7b8e6e --To-tag as121e4580 [Aug 18 10:33:58] DEBUG[13562] res_stasis_recording.c: 1629282837.155: Sending record(213008_uRBdlcrkyUhysKyMPjaNFREMlcwBMbYe.wav) command [Aug 18 10:33:58] DEBUG[13562] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:58] DEBUG[13635] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/record] with handler [httpstatus] len 10 [Aug 18 10:33:58] VERBOSE[13631] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[13631] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Initializing initreq for method INVITE - callid 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116992@178.62.121.41 SIP/2.0 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 3 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 6 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:58 GMT [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:58] DEBUG[13562] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13634] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel Recorder/ARI-0000001a;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:33:58] DEBUG[13634] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13626] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: alaw -> slin [Aug 18 10:33:58] DEBUG[13635] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/record] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13640] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13640] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:58] DEBUG[13635] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/record] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13635] http.c: Match made with [ari] [Aug 18 10:33:58] VERBOSE[13631] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[13640] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Finding handler for bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/record [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13640] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13640] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Finding handler for bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:58] DEBUG[13640] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:58] DEBUG[13640] stasis.c: Creating topic. name: bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a, detail: [Aug 18 10:33:58] DEBUG[13640] stasis.c: Topic 'bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a': 0x7f0c980a8850 created [Aug 18 10:33:58] DEBUG[13640] stasis.c: Creating topic. name: cache:203/bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a, detail: [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Aug 18 10:33:58] DEBUG[13631] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Finding handler for 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:58] VERBOSE[13631] dial.c: Called zvonobot/79821116992 [Aug 18 10:33:58] DEBUG[13637] http.c: HTTP opening session. Top level [Aug 18 10:33:58] DEBUG[13637] http.c: HTTP Request URI is /ari/channels/212977/snoop?app=calls_0&spy=in [Aug 18 10:33:58] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:33:58] DEBUG[13637] http.c: match request [ari/channels/212977/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:58] DEBUG[13637] http.c: match request [ari/channels/212977/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:58] DEBUG[13637] http.c: match request [ari/channels/212977/snoop] with handler [ari] len 3 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:58] DEBUG[13637] http.c: Match made with [ari] [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:58] VERBOSE[13632] app.c: x=0, open writing: /var/spool/asterisk/recording/212977_vtgwVPwmpVRoqoJmilBCHrdNgFSFbIMS format: wav, 0x7f0c8c06cac0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Stopping retransmission on '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:58] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:58] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:58] DEBUG[13640] stasis.c: Topic 'cache:203/bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a': 0x7f0c980a77c0 created [Aug 18 10:33:58] DEBUG[13638] app.c: play_and_record: , /var/spool/asterisk/recording/213008_uRBdlcrkyUhysKyMPjaNFREMlcwBMbYe, 'wav' [Aug 18 10:33:58] DEBUG[13638] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Finding handler for channels/212977/snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Finding handler for channels [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:58] VERBOSE[13638] app.c: x=0, open writing: /var/spool/asterisk/recording/213008_uRBdlcrkyUhysKyMPjaNFREMlcwBMbYe format: wav, 0x7f0c9c03dc70 [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:58] VERBOSE[13636] chan_sip.c: Audio is at 11378 [Aug 18 10:33:58] DEBUG[13573] stasis.c: Creating topic. name: channel:1629282838.171, detail: [Aug 18 10:33:58] DEBUG[13573] stasis.c: Topic 'channel:1629282838.171': 0x7f0c9c024d20 created [Aug 18 10:33:58] DEBUG[13573] stasis.c: Creating topic. name: cache:204/channel:1629282838.171, detail: [Aug 18 10:33:58] DEBUG[13573] stasis.c: Topic 'cache:204/channel:1629282838.171': 0x7f0c9c032730 created [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK683535b4 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Contact: Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:58] DEBUG[13635] res_ari.c: No explicit handler found for 357a4882-a24d-489f-8ff8-98badd81b2ee. Using wildcard bridgeId. [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13573] channel.c: Channel 0x7f0c9c024160 'Announcer/ARI-0000001d;1' allocated [Aug 18 10:33:58] DEBUG[13573] stasis.c: Creating topic. name: channel:1629282838.172, detail: [Aug 18 10:33:58] DEBUG[13573] stasis.c: Topic 'channel:1629282838.172': 0x7f0c9c001f00 created [Aug 18 10:33:58] DEBUG[13573] stasis.c: Creating topic. name: cache:205/channel:1629282838.172, detail: [Aug 18 10:33:58] DEBUG[13573] stasis.c: Topic 'cache:205/channel:1629282838.172': 0x7f0c9c044450 created [Aug 18 10:33:58] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] VERBOSE[13636] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:58] VERBOSE[13636] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:58] VERBOSE[13636] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Initializing initreq for method INVITE - callid 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116993@178.62.121.41 SIP/2.0 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 3 [ 52]: From: ;tag=as34e313e7 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 6 [ 60]: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:58 GMT [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:58] VERBOSE[13636] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [Aug 18 10:33:58] DEBUG[13636] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:58] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (1) INVITE - 5 [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:58] DEBUG[13557] channel.c: Channel 0x7f0c70070730 'Announcer/ARI-0000001b;1' allocated [Aug 18 10:33:58] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Finding handler for 212977 [Aug 18 10:33:58] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:58] DEBUG[13557] stasis.c: Creating topic. name: channel:1629282838.173, detail: [Aug 18 10:33:58] DEBUG[13557] stasis.c: Topic 'channel:1629282838.173': 0x7f0c7007cb00 created [Aug 18 10:33:58] DEBUG[13557] stasis.c: Creating topic. name: cache:206/channel:1629282838.173, detail: [Aug 18 10:33:58] DEBUG[13557] stasis.c: Topic 'cache:206/channel:1629282838.173': 0x7f0c7005b740 created [Aug 18 10:33:58] DEBUG[13232] res_rtp_asterisk.c: (0x7f0c3401c090) RTP audio difference is 688, ms is 106 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116997@178.62.121.41", nonce="2dbc5b08", response="c92db07f4aae82ff2863950559637b69" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485184 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (2) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116999@178.62.121.41", nonce="048e446a", response="229b027e812dcf7d91509c351fa56509" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #41 (3) INVITE - 5 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #41)) [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010121 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:58] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Finding handler for record [Aug 18 10:33:58] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 720, ms is 110 [Aug 18 10:33:58] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channels create: Didn't match 212977 [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channels externalMedia: Didn't match 212977 [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:58] DEBUG[13299] channel.c: Channel 0x7f0c9803f2e0 'Recorder/ARI-00000007;1' destroying [Aug 18 10:33:58] DEBUG[13637] res_ari.c: No explicit handler found for 212977. Using wildcard channelId. [Aug 18 10:33:58] DEBUG[13614] channel.c: Channel 0x7f0c2c090fb0 'SIP/zvonobot-00000052' allocated [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:58] DEBUG[13614] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:58] DEBUG[13635] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:58] DEBUG[13635] stasis.c: Creating topic. name: channel:1629282838.174, detail: [Aug 18 10:33:58] DEBUG[13635] stasis.c: Topic 'channel:1629282838.174': 0x7f0c94064c70 created [Aug 18 10:33:58] DEBUG[13635] stasis.c: Creating topic. name: cache:207/channel:1629282838.174, detail: [Aug 18 10:33:58] DEBUG[13635] stasis.c: Topic 'cache:207/channel:1629282838.174': 0x7f0c940682b0 created [Aug 18 10:33:58] DEBUG[13640] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' can not use native RTP bridge as two channels are required [Aug 18 10:33:58] DEBUG[13640] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:58] DEBUG[13640] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:58] DEBUG[13640] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:58] DEBUG[13640] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:58] DEBUG[13640] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: calling simple_bridge technology constructor [Aug 18 10:33:58] DEBUG[13640] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: calling simple_bridge technology start [Aug 18 10:33:58] DEBUG[13640] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13299] stasis.c: Destroying topic. name: cache:97/channel:1629282832.83, detail: [Aug 18 10:33:58] DEBUG[13299] stasis.c: Topic 'cache:97/channel:1629282832.83': 0x7f0c98022fd0 destroyed [Aug 18 10:33:58] DEBUG[13299] stasis.c: Destroying topic. name: channel:1629282832.83, detail: [Aug 18 10:33:58] DEBUG[13299] stasis.c: Topic 'channel:1629282832.83': 0x7f0c98034880 destroyed [Aug 18 10:33:58] DEBUG[13640] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Finding handler for snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:58] VERBOSE[13636] dial.c: Called zvonobot/79821116993 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:58] DEBUG[13532] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:58] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:58] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:58] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:58] DEBUG[13637] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:58] DEBUG[13642] http.c: HTTP opening session. Top level [Aug 18 10:33:58] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:58] DEBUG[13614] res_stasis.c: calls_0: Subscribing to 213053 [Aug 18 10:33:58] DEBUG[13602] channel.c: Channel 0x2c6e150 'SIP/zvonobot-00000053' allocated [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:58] DEBUG[13602] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:58] DEBUG[13614] stasis/app.c: Channel '213053' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[13614] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13602] res_stasis.c: calls_0: Subscribing to 213051 [Aug 18 10:33:58] DEBUG[13614] http.c: HTTP closing session. Top level [Aug 18 10:33:58] DEBUG[13602] stasis/app.c: Channel '213051' is 1 interested in calls_0 [Aug 18 10:33:58] DEBUG[13602] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:58] DEBUG[13602] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Outgoing Call for 79821116989 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[13645] chan_sip.c: Audio is at 15904 [Aug 18 10:33:59] VERBOSE[13645] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13645] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13645] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Initializing initreq for method INVITE - callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116989@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 3 [ 52]: From: ;tag=as79336d5f [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Outgoing Call for 79821116987 [Aug 18 10:33:58] DEBUG[13642] http.c: HTTP Request URI is /ari/channels/213008/snoop?app=calls_0&spy=in [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 6 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13642] http.c: match request [ari/channels/213008/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[13644] chan_sip.c: Audio is at 14624 [Aug 18 10:33:59] VERBOSE[13644] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13644] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13644] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13642] http.c: match request [ari/channels/213008/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13642] http.c: match request [ari/channels/213008/snoop] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13642] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Finding handler for channels/213008/snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Finding handler for 213008 [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channels create: Didn't match 213008 [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channels externalMedia: Didn't match 213008 [Aug 18 10:33:59] DEBUG[13642] res_ari.c: No explicit handler found for 213008. Using wildcard channelId. [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Finding handler for snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:33:59] DEBUG[13642] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:33:59] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13532] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:59] DEBUG[13532] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:59] DEBUG[13532] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:33:59] DEBUG[13532] channel.c: Channel Announcer/ARI-00000017;1 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[13532] channel.c: Channel 0x7f0c1808d1d0 'Announcer/ARI-00000017;1' hanging up. Refs: 2 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 From: ;tag=as30c9fc1b To: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116997@178.62.121.41", nonce="2dbc5b08", response="c92db07f4aae82ff2863950559637b69" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485184 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13645] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:33:59] DEBUG[13645] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0;received=159.65.48.104 From: ;tag=as6093d024 To: ;tag=as49ef3a53 Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ce9240" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as49ef3a53 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ce9240" [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag as49ef3a53 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #47 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0 Max-Forwards: 70 From: ;tag=as6093d024 To: ;tag=as49ef3a53 Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Audio is at 13556 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] VERBOSE[13645] dial.c: Called zvonobot/79821116989 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Initializing initreq for method INVITE - callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116987@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 3 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 6 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13644] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #44 [Aug 18 10:33:59] DEBUG[13644] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13247] channel.c: Channel 0x7f0c9802bfd0 'Announcer/ARI-00000006;1' destroying [Aug 18 10:33:59] DEBUG[13589] channel.c: Channel 0x7f0cb00de970 'SIP/zvonobot-00000054' allocated [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13589] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] DEBUG[13589] res_stasis.c: calls_0: Subscribing to 213044 [Aug 18 10:33:59] DEBUG[13589] stasis/app.c: Channel '213044' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13247] stasis.c: Destroying topic. name: cache:83/channel:1629282831.70, detail: [Aug 18 10:33:59] DEBUG[13247] stasis.c: Topic 'cache:83/channel:1629282831.70': 0x7f0c98023d90 destroyed [Aug 18 10:33:59] DEBUG[13247] stasis.c: Destroying topic. name: channel:1629282831.70, detail: [Aug 18 10:33:59] DEBUG[13247] stasis.c: Topic 'channel:1629282831.70': 0x7f0c9802a530 destroyed [Aug 18 10:33:59] DEBUG[13302] channel.c: Channel 0x7f0c800219b0 'SIP/zvonobot-0000001b' destroying [Aug 18 10:33:59] DEBUG[13245] bridge_channel.c: Setting 0x7f0c9802b450(Announcer/ARI-00000006;2) state from:0 to:1 [Aug 18 10:33:59] DEBUG[13589] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13589] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13302] channel.c: Channel 0x7f0cac03e430 'Snoop/212993-00000003' destroying [Aug 18 10:33:59] DEBUG[13608] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20' [Aug 18 10:33:59] DEBUG[13608] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:59] DEBUG[13608] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Outgoing Call for 79821116996 [Aug 18 10:33:59] DEBUG[13649] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282839.175, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.175': 0x7f0c300dca50 created [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: cache:208/channel:1629282839.175, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:208/channel:1629282839.175': 0x7f0c300a8980 created [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:33:59] DEBUG[13302] stasis.c: Destroying topic. name: cache:100/channel:1629282832.85, detail: [Aug 18 10:33:59] DEBUG[13245] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pulling 0x7f0c9802b450(Announcer/ARI-00000006;2) [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:33:59] VERBOSE[13245] bridge_channel.c: Channel Announcer/ARI-00000006;2 left 'softmix' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:59] DEBUG[13649] http.c: HTTP Request URI is /ari/channels/212993 [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel '212993': is 0 interested in calls_0 [Aug 18 10:33:59] DEBUG[13302] stasis.c: Topic 'cache:100/channel:1629282832.85': 0x7f0cac01cfb0 destroyed [Aug 18 10:33:59] DEBUG[13302] stasis.c: Destroying topic. name: channel:1629282832.85, detail: [Aug 18 10:33:59] DEBUG[13302] stasis.c: Topic 'channel:1629282832.85': 0x7f0cac01e180 destroyed [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel '212993' unsubscribed from calls_0 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[13648] chan_sip.c: Audio is at 10612 [Aug 18 10:33:59] VERBOSE[13648] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13648] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13648] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13319] bridge_channel.c: Setting 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) state from:0 to:1 [Aug 18 10:33:59] DEBUG[13245] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c9802b450(Announcer/ARI-00000006;2) is leaving softmix technology [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: cache:34/channel:212993, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'cache:34/channel:212993': 0x7f0c80024280 destroyed [Aug 18 10:33:59] DEBUG[13319] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: pulling 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) [Aug 18 10:33:59] VERBOSE[13319] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50409-0x7f0c10045b20 left 'simple_bridge' stasis-bridge <9d1bf1e2-893f-4249-b006-4b3a345e76a2> [Aug 18 10:33:59] DEBUG[13319] bridge_channel.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) is leaving simple_bridge technology [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: cache:208/channel:1629282839.175, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:208/channel:1629282839.175': 0x7f0c300a8980 destroyed [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282839.175, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.175': 0x7f0c300dca50 destroyed [Aug 18 10:33:59] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: channel:212993, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'channel:212993': 0x7f0c80024080 destroyed [Aug 18 10:33:59] DEBUG[13649] http.c: match request [ari/channels/212993] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001b', '', 'Stasis', 'calls_0', 11, 5, 'ANSWERED', 3, '', '212993', '')] [Aug 18 10:33:59] DEBUG[13649] http.c: match request [ari/channels/212993] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282839.176, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.176': 0x7f0c300dca50 created [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: cache:209/channel:1629282839.176, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:209/channel:1629282839.176': 0x7f0c300a5fd0 created [Aug 18 10:33:59] DEBUG[13245] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2'. Checking compatability for channels 'SIP/zvonobot-00000016' and 'Recorder/ARI-00000005;2' [Aug 18 10:33:59] DEBUG[13245] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as channel 'SIP/zvonobot-00000016' has features which prevent it [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13245] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] VERBOSE[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: switching from softmix technology to simple_bridge [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: moving 0x7f0c30064e70(SIP/zvonobot-00000016) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: moving 0x7f0c8c037dd0(Recorder/ARI-00000005;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling softmix technology stop [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel Recorder/ARI-00000005;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel Recorder/ARI-00000005;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel Recorder/ARI-00000005;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel Recorder/ARI-00000005;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: deferring softmix technology destructor [Aug 18 10:33:59] DEBUG[13245] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: queueing action type:13 sub:1000 [Aug 18 10:33:59] DEBUG[13649] http.c: match request [ari/channels/212993] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13319] bridge_native_rtp.c: Bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13319] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13319] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13319] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13319] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13319] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2 is already using the new technology. [Aug 18 10:33:59] DEBUG[13319] stasis/control.c: robot_212993, 9d1bf1e2-893f-4249-b006-4b3a345e76a2: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[13319] stasis/app.c: bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2': is 2 interested in calls_0 [Aug 18 10:33:59] DEBUG[13319] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[13312] stasis/control.c: robot_212993: Channel departing bridge [Aug 18 10:33:59] DEBUG[13312] bridge.c: Waiting for 0x7f0c24006440(UnicastRTP/127.0.0.1:50409-0x7f0c10045b20) bridge thread to die. [Aug 18 10:33:59] DEBUG[13312] stasis/app.c: channel 'robot_212993': is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13312] channel.c: Channel 0x7f0c1004e480 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20' hanging up. Refs: 2 [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: cache:209/channel:1629282839.176, detail: [Aug 18 10:33:59] DEBUG[13649] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:209/channel:1629282839.176': 0x7f0c300a5fd0 destroyed [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282839.176, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.176': 0x7f0c300dca50 destroyed [Aug 18 10:33:59] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <>', '', 's', 'default', 'Snoop/212993-00000003', 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20', 'Stasis', 'calls_0', 6, 6, 'ANSWERED', 3, '', '1629282832.85', '')] [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:33:59] VERBOSE[13644] dial.c: Called zvonobot/79821116987 [Aug 18 10:33:59] DEBUG[13232] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-00000016' [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Finding handler for channels/212993 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Initializing initreq for method INVITE - callid 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116996@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 3 [ 52]: From: ;tag=as1dc5ead1 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 From: ;tag=as686a751a To: ;tag=as6e8cb5a3 Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1721442823 1721442823 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11280 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 6 [ 60]: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13648] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Aug 18 10:33:59] DEBUG[13648] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as686a751a [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Finding handler for channels [Aug 18 10:33:59] VERBOSE[13648] dial.c: Called zvonobot/79821116996 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e8cb5a3 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1721442823 1721442823 IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11280 RTP/AVP 0 8 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 (Checking To) --From tag as686a751a --To-tag as6e8cb5a3 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Finding handler for 212993 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking channels create: Didn't match 212993 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13649] res_ari.c: Checking channels externalMedia: Didn't match 212993 [Aug 18 10:33:59] DEBUG[13649] res_ari.c: No explicit handler found for 212993. Using wildcard channelId. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '398559732fb8625271bea90231b90490@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Got SDP version 1721442823 and unique parts [root 1721442823 IN IP4 178.62.121.41] [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1721442823 1721442823 IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:59] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP creating BEGIN DTMF Frame: 54 (6), at 178.62.121.41:14674 [Aug 18 10:33:59] DTMF[13556] channel.c: DTMF begin '6' received on SIP/zvonobot-0000002b [Aug 18 10:33:59] DTMF[13556] channel.c: DTMF begin passthrough '6' on SIP/zvonobot-0000002b [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:59] DTMF[13638] channel.c: DTMF begin '6' received on Recorder/ARI-0000001a;1 [Aug 18 10:33:59] DTMF[13638] channel.c: DTMF begin passthrough '6' on Recorder/ARI-0000001a;1 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] DEBUG[13613] channel.c: Channel 0x7f0c20081870 'SIP/zvonobot-00000056' allocated [Aug 18 10:33:59] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13613] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:59] DEBUG[13613] res_stasis.c: calls_0: Subscribing to 213052 [Aug 18 10:33:59] DEBUG[13613] stasis/app.c: Channel '213052' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Outgoing Call for 79821116988 [Aug 18 10:33:59] DEBUG[13613] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13613] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[13655] chan_sip.c: Audio is at 14750 [Aug 18 10:33:59] VERBOSE[13655] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13655] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13655] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] DEBUG[20534] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:59] DEBUG[13234] channel.c: Recorder/ARI-00000005;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:59] DEBUG[13246] bridge_softmix.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: stopping mixing thread [Aug 18 10:33:59] DEBUG[20534] bridge_softmix.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: Waiting for mixing thread to die. [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Initializing initreq for method INVITE - callid 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116988@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13232] channel.c: SIP/zvonobot-00000016: Dropping redundant connected line update "" <>. [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 3 [ 52]: From: ;tag=as02885f54 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 6 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13655] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #49 [Aug 18 10:33:59] DEBUG[13655] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel 0x7f0c98034ab0 'Announcer/ARI-00000006;2' hanging up. Refs: 2 [Aug 18 10:33:59] VERBOSE[13655] dial.c: Called zvonobot/79821116988 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) ICE set role failed; no ice instance [Aug 18 10:33:59] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) RTCP setting address on RTP instance [Aug 18 10:33:59] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c74011cd0 -- Strict RTP learning after remote address set to: 178.62.121.41:11280 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:11280 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb01025a8) from 0x7f0c147e2330 to 0x7f0c74010768 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb006a808) from 0x7f0c147e2330 to 0x7f0c74010768 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00e3d08) from 0x7f0c147e2330 to 0x7f0c74010768 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) RTCP ignoring duplicate property [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002a setting read format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002a setting write format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS - ast_rtp_activate rtp=0x7f0c74011cd0 - setup and perform DTLS' [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74011cd0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74011cd0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Strict routing enforced for session 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117033@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c130273 Max-Forwards: 70 From: ;tag=as686a751a To: ;tag=as6e8cb5a3 Contact: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117001@178.62.121.41", nonce="0346307c", response="cf82a81f12417d8ec0d755f08d1960a5" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 629437933 629437934 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10382 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117006@178.62.121.41", nonce="2e114138", response="546d7b2d9fa7f2afdf06a7c01e9051cf" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379123 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[13177] dial.c: SIP/zvonobot-0000002a answered [Aug 18 10:33:59] VERBOSE[13177] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000002a [Aug 18 10:33:59] DEBUG[13177] stasis/app.c: Channel '213007' is 2 interested in calls_0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13501] channel.c: Channel 0x7f0c10068200 'Announcer/ARI-00000015;1' destroying [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13610] channel.c: Channel 0x7f0c180d2b30 'SIP/zvonobot-00000057' allocated [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13610] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:59] DEBUG[13496] bridge_channel.c: Setting 0x7f0c1004ffb0(Announcer/ARI-00000015;2) state from:0 to:1 [Aug 18 10:33:59] VERBOSE[13177] res_rtp_asterisk.c: 0x7f0c74011cd0 -- Strict RTP switching to RTP target address 178.62.121.41:11280 as source [Aug 18 10:33:59] DEBUG[13177] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:59] DEBUG[13177] channel.c: Channel SIP/zvonobot-0000002a setting read format path: ulaw -> alaw [Aug 18 10:33:59] DEBUG[13177] channel.c: Channel SIP/zvonobot-0000002a setting write format path: alaw -> ulaw [Aug 18 10:33:59] DEBUG[13501] stasis.c: Destroying topic. name: cache:161/channel:1629282835.134, detail: [Aug 18 10:33:59] DEBUG[13501] stasis.c: Topic 'cache:161/channel:1629282835.134': 0x7f0c10065140 destroyed [Aug 18 10:33:59] DEBUG[13501] stasis.c: Destroying topic. name: channel:1629282835.134, detail: [Aug 18 10:33:59] DEBUG[13501] stasis.c: Topic 'channel:1629282835.134': 0x7f0c100699c0 destroyed [Aug 18 10:33:59] DEBUG[13496] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pulling 0x7f0c1004ffb0(Announcer/ARI-00000015;2) [Aug 18 10:33:59] VERBOSE[13496] bridge_channel.c: Channel Announcer/ARI-00000015;2 left 'softmix' stasis-bridge [Aug 18 10:33:59] DEBUG[13496] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c1004ffb0(Announcer/ARI-00000015;2) is leaving softmix technology [Aug 18 10:33:59] DEBUG[13657] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[13657] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13628] channel.c: Channel 0x7f0c7806bc10 'Recorder/ARI-0000001c;1' allocated [Aug 18 10:33:59] DEBUG[13628] stasis.c: Creating topic. name: channel:1629282839.177, detail: [Aug 18 10:33:59] DEBUG[13628] stasis.c: Topic 'channel:1629282839.177': 0x7f0c78074f80 created [Aug 18 10:33:59] DEBUG[13628] stasis.c: Creating topic. name: cache:210/channel:1629282839.177, detail: [Aug 18 10:33:59] DEBUG[13628] stasis.c: Topic 'cache:210/channel:1629282839.177': 0x7f0c7806aed0 created [Aug 18 10:33:59] DEBUG[13592] channel.c: Channel 0x7f0cac079f80 'SIP/zvonobot-00000055' allocated [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13592] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] DEBUG[13496] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267'. Checking compatability for channels 'SIP/zvonobot-0000002e' and 'Recorder/ARI-00000014;2' [Aug 18 10:33:59] DEBUG[13496] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as channel 'SIP/zvonobot-0000002e' has features which prevent it [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13496] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] VERBOSE[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: switching from softmix technology to simple_bridge [Aug 18 10:33:59] DEBUG[13657] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:33:59] DEBUG[13657] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13657] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13657] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c9006b170(SIP/zvonobot-0000002e) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116999@178.62.121.41", nonce="048e446a", response="229b027e812dcf7d91509c351fa56509" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c8808e340(Recorder/ARI-00000014;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[13657] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology stop [Aug 18 10:33:59] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP creating END DTMF Frame: 54 (6), at 178.62.121.41:14674 [Aug 18 10:33:59] DTMF[13556] channel.c: DTMF end '6' received on SIP/zvonobot-0000002b, duration 100 ms [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session timer started: 54 - 398559732fb8625271bea90231b90490@159.65.48.104:5060 1768000ms [Aug 18 10:33:59] DEBUG[13657] stasis.c: Creating topic. name: bridge:aba705f1-c39f-408a-8a02-8c7f66ee7c7d, detail: [Aug 18 10:33:59] DTMF[13556] channel.c: DTMF end accepted with begin '6' on SIP/zvonobot-0000002b [Aug 18 10:33:59] DTMF[13556] channel.c: DTMF end passthrough '6' on SIP/zvonobot-0000002b [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is joining simple_bridge technology [Aug 18 10:33:59] DTMF[13638] channel.c: DTMF end '6' received on Recorder/ARI-0000001a;1, duration 100 ms [Aug 18 10:33:59] DTMF[13638] channel.c: DTMF end accepted with begin '6' on Recorder/ARI-0000001a;1 [Aug 18 10:33:59] DTMF[13638] channel.c: DTMF end passthrough '6' on Recorder/ARI-0000001a;1 [Aug 18 10:33:59] DEBUG[13616] channel.c: Channel 0x7f0c28104e60 'SIP/zvonobot-00000058' allocated [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13616] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 494 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 494 [Aug 18 10:33:59] DEBUG[13657] stasis.c: Topic 'bridge:aba705f1-c39f-408a-8a02-8c7f66ee7c7d': 0x7f0c100649c0 created [Aug 18 10:33:59] DEBUG[13616] res_stasis.c: calls_0: Subscribing to 213049 [Aug 18 10:33:59] DEBUG[13616] stasis/app.c: Channel '213049' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13610] res_stasis.c: calls_0: Subscribing to 213045 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Outgoing Call for 79821116991 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 From: ;tag=as2b432b30 To: ;tag=as069541b6 Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0346307c" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as069541b6 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel Recorder/ARI-00000014;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13610] stasis/app.c: Channel '213045' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel Recorder/ARI-00000014;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13610] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: deferring softmix technology destructor [Aug 18 10:33:59] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13616] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13610] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13496] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: queueing action type:13 sub:1000 [Aug 18 10:33:59] DEBUG[13616] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13657] stasis.c: Creating topic. name: cache:211/bridge:aba705f1-c39f-408a-8a02-8c7f66ee7c7d, detail: [Aug 18 10:33:59] DEBUG[13657] stasis.c: Topic 'cache:211/bridge:aba705f1-c39f-408a-8a02-8c7f66ee7c7d': 0x7f0c100699c0 created [Aug 18 10:33:59] DEBUG[13592] res_stasis.c: calls_0: Subscribing to 213046 [Aug 18 10:33:59] DEBUG[13592] stasis/app.c: Channel '213046' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13592] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13592] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0346307c" [Aug 18 10:33:59] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] VERBOSE[13658] chan_sip.c: Audio is at 15836 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag as069541b6 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Outgoing Call for 79821116995 [Aug 18 10:33:59] DEBUG[13657] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13657] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13657] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13657] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] VERBOSE[13658] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13658] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13658] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Outgoing Call for 79821116994 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[13657] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13657] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[13657] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13468] channel.c: Recorder/ARI-00000014;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:59] DEBUG[13499] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: stopping mixing thread [Aug 18 10:33:59] DEBUG[20534] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:59] DEBUG[20534] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: Waiting for mixing thread to die. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Initializing initreq for method INVITE - callid 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13657] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13657] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13459] channel.c: SIP/zvonobot-0000002e: Dropping redundant connected line update "" <>. [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel 0x7f0c1006f850 'Announcer/ARI-00000015;2' hanging up. Refs: 2 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116991@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 3 [ 52]: From: ;tag=as3f810040 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 6 [ 60]: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13658] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #55 [Aug 18 10:33:59] DEBUG[13658] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] VERBOSE[13661] chan_sip.c: Audio is at 17196 [Aug 18 10:33:59] VERBOSE[13661] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[13661] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (2) INVITE - 5 [Aug 18 10:33:59] VERBOSE[13661] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] VERBOSE[13658] dial.c: Called zvonobot/79821116991 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Initializing initreq for method INVITE - callid 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116994@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13662] http.c: HTTP opening session. Top level [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[13662] http.c: HTTP Request URI is /ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/addChannel?channel=213007 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 3 [ 52]: From: ;tag=as2618d3a5 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 6 [ 60]: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[13659] chan_sip.c: Audio is at 13804 [Aug 18 10:33:59] DEBUG[13662] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13662] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16;received=159.65.48.104 From: ;tag=as3edf3f1c To: ;tag=as177609bc Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e114138" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0343aa16;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as177609bc [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] VERBOSE[13659] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13659] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13659] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Initializing initreq for method INVITE - callid 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116995@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 3 [ 52]: From: ;tag=as6d7500c6 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 6 [ 60]: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e114138" [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag as177609bc [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13661] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #60 [Aug 18 10:33:59] DEBUG[13661] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:59] VERBOSE[13661] dial.c: Called zvonobot/79821116994 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116997@178.62.121.41", nonce="2dbc5b08", response="c92db07f4aae82ff2863950559637b69" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485184 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce;received=159.65.48.104 From: ;tag=as52d5bd88 To: ;tag=as171b84c8 Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1cff09" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as171b84c8 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1cff09" [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag as171b84c8 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #41 [Aug 18 10:33:59] DEBUG[13662] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/addChannel] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13662] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Finding handler for bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/addChannel [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Finding handler for aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13662] res_ari.c: No explicit handler found for aba705f1-c39f-408a-8a02-8c7f66ee7c7d. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Finding handler for addChannel [Aug 18 10:33:59] DEBUG[13662] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:59] DEBUG[13662] stasis/control.c: 213007: Sending channel add_to_bridge command [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:33:59] DEBUG[13177] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000002a [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:33:59] DEBUG[13177] stasis/control.c: 213007: Adding to bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:33:59] DEBUG[13177] stasis/app.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13666] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is joining [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce Max-Forwards: 70 From: ;tag=as52d5bd88 To: ;tag=as171b84c8 Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Audio is at 11140 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #63 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13666] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pushing 0x7f0c7006de00(SIP/zvonobot-0000002a) [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13659] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #64 [Aug 18 10:33:59] DEBUG[13659] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[13659] dial.c: Called zvonobot/79821116995 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] VERBOSE[13666] bridge_channel.c: Channel SIP/zvonobot-0000002a joined 'simple_bridge' stasis-bridge [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 From: ;tag=as73737e94 To: ;tag=as6e22f1d1 Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 870064292 870064292 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16540 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as73737e94 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e22f1d1 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 870064292 870064292 IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16540 RTP/AVP 0 8 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 (Checking To) --From tag as73737e94 --To-tag as6e22f1d1 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Got SDP version 870064292 and unique parts [root 870064292 IN IP4 178.62.121.41] [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 870064292 870064292 IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800cb50) ICE set role failed; no ice instance [Aug 18 10:33:59] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP setting address on RTP instance [Aug 18 10:33:59] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0ca8020ff0 -- Strict RTP learning after remote address set to: 178.62.121.41:16540 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:16540 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0036c68) from 0x7f0c147e2330 to 0x7f0ca800cd28 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb006c668) from 0x7f0c147e2330 to 0x7f0ca800cd28 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0098688) from 0x7f0c147e2330 to 0x7f0ca800cd28 [Aug 18 10:33:59] DEBUG[13666] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13666] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP ignoring duplicate property [Aug 18 10:33:59] DEBUG[13666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:59] DEBUG[13666] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13666] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002f setting read format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002f setting write format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[13666] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca800cb50) DTLS - ast_rtp_activate rtp=0x7f0ca8020ff0 - setup and perform DTLS' [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8020ff0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8020ff0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:59] DEBUG[13666] res_rtp_asterisk.c: (0x7f0c74010590) RTP changing ssrc from 152795812 to 714453239 due to a source change [Aug 18 10:33:59] DEBUG[13177] stasis/app.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' is 2 interested in calls_0 [Aug 18 10:33:59] DEBUG[13668] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13662] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:59] DEBUG[13662] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13668] http.c: HTTP Request URI is /ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/record?name=213007_pPaTJtaVJwuRGbJSRCzOzmloUbPCTlFs&format=wav [Aug 18 10:33:59] DEBUG[13618] channel.c: Channel 0x7f0c340bc830 'SIP/zvonobot-00000059' allocated [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:33:59] DEBUG[13618] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:33:59] DEBUG[13618] res_stasis.c: calls_0: Subscribing to 213050 [Aug 18 10:33:59] DEBUG[13618] stasis/app.c: Channel '213050' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:59] DEBUG[13668] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/record] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13532] channel.c: Channel 0x7f0c1808d1d0 'Announcer/ARI-00000017;1' destroying [Aug 18 10:33:59] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:59] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Recorder - ARI [Aug 18 10:33:59] DEBUG[13573] channel.c: Channel 0x7f0c9c05ac20 'Announcer/ARI-0000001d;2' allocated [Aug 18 10:33:59] DEBUG[13573] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] DEBUG[13573] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001d;1' [Aug 18 10:33:59] DEBUG[13668] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/record] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13618] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13532] stasis.c: Destroying topic. name: cache:164/channel:1629282835.137, detail: [Aug 18 10:33:59] DEBUG[13532] stasis.c: Topic 'cache:164/channel:1629282835.137': 0x7f0c18090df0 destroyed [Aug 18 10:33:59] DEBUG[13532] stasis.c: Destroying topic. name: channel:1629282835.137, detail: [Aug 18 10:33:59] DEBUG[13532] stasis.c: Topic 'channel:1629282835.137': 0x7f0c18093d90 destroyed [Aug 18 10:33:59] DEBUG[13668] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/record] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Outgoing Call for 79821116990 [Aug 18 10:33:59] DEBUG[13498] bridge_channel.c: Setting 0x7f0c180a5170(Announcer/ARI-00000017;2) state from:0 to:1 [Aug 18 10:33:59] DEBUG[13668] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13618] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13557] channel.c: Channel 0x7f0c7007bb40 'Announcer/ARI-0000001b;2' allocated [Aug 18 10:33:59] DEBUG[13557] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] DEBUG[13557] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001b;1' [Aug 18 10:33:59] DEBUG[13635] channel.c: Channel 0x7f0c94066640 'Recorder/ARI-0000001e;1' allocated [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] DEBUG[13635] stasis.c: Creating topic. name: channel:1629282839.178, detail: [Aug 18 10:33:59] DEBUG[13672] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) is joining [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] DEBUG[13635] stasis.c: Topic 'channel:1629282839.178': 0x7f0c9405fd20 created [Aug 18 10:33:59] DEBUG[13635] stasis.c: Creating topic. name: cache:212/channel:1629282839.178, detail: [Aug 18 10:33:59] DEBUG[13635] stasis.c: Topic 'cache:212/channel:1629282839.178': 0x7f0c9405db50 created [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:33:59] DEBUG[13671] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9c048790(Announcer/ARI-0000001d;2) is joining [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:33:59] VERBOSE[13669] chan_sip.c: Audio is at 10086 [Aug 18 10:33:59] VERBOSE[13669] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:33:59] VERBOSE[13669] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:33:59] VERBOSE[13669] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Initializing initreq for method INVITE - callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116990@178.62.121.41 SIP/2.0 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 3 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 6 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:33:59 GMT [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Strict routing enforced for session 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Finding handler for bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/record [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[13671] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pushing 0x7f0c9c048790(Announcer/ARI-0000001d;2) [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13498] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pulling 0x7f0c180a5170(Announcer/ARI-00000017;2) [Aug 18 10:33:59] VERBOSE[13498] bridge_channel.c: Channel Announcer/ARI-00000017;2 left 'softmix' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:33:59] DEBUG[13498] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c180a5170(Announcer/ARI-00000017;2) is leaving softmix technology [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] VERBOSE[13669] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #57 [Aug 18 10:33:59] DEBUG[13669] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117031@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0ce2b2fc Max-Forwards: 70 From: ;tag=as73737e94 To: ;tag=as6e22f1d1 Contact: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[13198] dial.c: SIP/zvonobot-0000002f answered [Aug 18 10:33:59] VERBOSE[13669] dial.c: Called zvonobot/79821116990 [Aug 18 10:33:59] DEBUG[13672] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pushing 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:33:59] DEBUG[13672] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] VERBOSE[13672] bridge_channel.c: Channel Announcer/ARI-0000001b;2 joined 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (2) INVITE - 5 [Aug 18 10:33:59] VERBOSE[13198] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000002f [Aug 18 10:33:59] DEBUG[13198] stasis/app.c: Channel '213009' is 2 interested in calls_0 [Aug 18 10:33:59] VERBOSE[13198] res_rtp_asterisk.c: 0x7f0ca8020ff0 -- Strict RTP switching to RTP target address 178.62.121.41:16540 as source [Aug 18 10:33:59] DEBUG[13198] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:59] DEBUG[13198] channel.c: Channel SIP/zvonobot-0000002f setting read format path: ulaw -> alaw [Aug 18 10:33:59] DEBUG[13198] channel.c: Channel SIP/zvonobot-0000002f setting write format path: alaw -> ulaw [Aug 18 10:33:59] DEBUG[13611] stasis.c: Creating topic. name: channel:1629282839.179, detail: [Aug 18 10:33:59] DEBUG[13611] stasis.c: Topic 'channel:1629282839.179': 0x7f0c240facd0 created [Aug 18 10:33:59] DEBUG[13611] stasis.c: Creating topic. name: cache:213/channel:1629282839.179, detail: [Aug 18 10:33:59] DEBUG[13611] stasis.c: Topic 'cache:213/channel:1629282839.179': 0x7f0c240f8a40 created [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13312] res_rtp_asterisk.c: (0x7f0c10045b20) DTLS stop [Aug 18 10:33:59] DEBUG[13312] res_rtp_asterisk.c: (0x7f0c10045b20) DTLS srtp - stopped timeout timer' [Aug 18 10:33:59] DEBUG[13312] res_rtp_asterisk.c: (0x7f0c10045b20) ICE RTP transport deallocating [Aug 18 10:33:59] DEBUG[13312] res_rtp_asterisk.c: (0x7f0c10045b20) ICE stopped [Aug 18 10:33:59] DEBUG[13312] rtp_engine.c: Destroyed RTP instance '0x7f0c10045b20' [Aug 18 10:33:59] DEBUG[13312] channel.c: Channel 0x7f0c1004e480 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20' destroying [Aug 18 10:33:59] DEBUG[13498] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162'. Checking compatability for channels 'SIP/zvonobot-00000023' and 'Recorder/ARI-00000013;2' [Aug 18 10:33:59] DEBUG[13498] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as channel 'SIP/zvonobot-00000023' has features which prevent it [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13245] channel.c: Channel 0x7f0c98034ab0 'Announcer/ARI-00000006;2' destroying [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13498] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] VERBOSE[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: switching from softmix technology to simple_bridge [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0ca0053060(SIP/zvonobot-00000023) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0c7c018d60(Recorder/ARI-00000013;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is leaving softmix technology (dummy) [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology stop [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282839.180, detail: [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Finding handler for aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.180': 0x7f0c300bc260 created [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: cache:214/channel:1629282839.180, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:214/channel:1629282839.180': 0x7f0c300bcc30 created [Aug 18 10:33:59] DEBUG[13674] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel Recorder/ARI-00000013;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13672] bridge.c: Chose bridge technology softmix [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel 'robot_212993': is 0 interested in calls_0 [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel 'robot_212993' unsubscribed from calls_0 [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: cache:103/channel:robot_212993, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'cache:103/channel:robot_212993': 0x7f0c1004fc00 destroyed [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212993, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'channel:robot_212993': 0x7f0c1004f9f0 destroyed [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: cache:214/channel:1629282839.180, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:214/channel:1629282839.180': 0x7f0c300bcc30 destroyed [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282839.180, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.180': 0x7f0c300bc260 destroyed [Aug 18 10:33:59] DEBUG[13674] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:59] VERBOSE[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: switching from simple_bridge technology to softmix [Aug 18 10:33:59] DEBUG[13245] stasis.c: Destroying topic. name: cache:84/channel:1629282831.71, detail: [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel Recorder/ARI-00000013;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13668] res_ari.c: No explicit handler found for aba705f1-c39f-408a-8a02-8c7f66ee7c7d. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology constructor [Aug 18 10:33:59] DEBUG[13245] stasis.c: Topic 'cache:84/channel:1629282831.71': 0x7f0c98037660 destroyed [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13674] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50409-0x7f0c10045b20', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212993', '')] [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c88050c90(SIP/zvonobot-00000020) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Finding handler for record [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: deferring softmix technology destructor [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c180a6470(Recorder/ARI-00000011;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13245] stasis.c: Destroying topic. name: channel:1629282831.71, detail: [Aug 18 10:33:59] DEBUG[13498] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: queueing action type:13 sub:1000 [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:33:59] DEBUG[13674] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13674] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13630] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720' [Aug 18 10:33:59] DEBUG[13245] stasis.c: Topic 'channel:1629282831.71': 0x7f0c98036be0 destroyed [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology stop [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13496] channel.c: Channel 0x7f0c1006f850 'Announcer/ARI-00000015;2' destroying [Aug 18 10:33:59] DEBUG[13496] stasis.c: Destroying topic. name: cache:166/channel:1629282835.139, detail: [Aug 18 10:33:59] DEBUG[13496] stasis.c: Topic 'cache:166/channel:1629282835.139': 0x7f0c100712f0 destroyed [Aug 18 10:33:59] DEBUG[13496] stasis.c: Destroying topic. name: channel:1629282835.139, detail: [Aug 18 10:33:59] DEBUG[13496] stasis.c: Topic 'channel:1629282835.139': 0x7f0c1005db30 destroyed [Aug 18 10:33:59] DEBUG[13630] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:33:59] DEBUG[13628] channel.c: Channel 0x7f0c78059c80 'Recorder/ARI-0000001c;2' allocated [Aug 18 10:33:59] DEBUG[13630] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13251] bridge_channel.c: Setting 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) state from:0 to:1 [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:33:59] DEBUG[13628] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] DEBUG[13251] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: pulling 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) [Aug 18 10:33:59] DEBUG[13668] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session timer started: 41 - 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 1768000ms [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244;received=159.65.48.104 From: ;tag=as6ceaa437 To: ;tag=as17140454 Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="048e446a" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK679a6244;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as17140454 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="048e446a" [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag as17140454 [Aug 18 10:33:59] VERBOSE[13251] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50291-0x7f0ca0023720 left 'simple_bridge' stasis-bridge <8b092052-108a-4921-8aad-1aecb4e2c824> [Aug 18 10:33:59] DEBUG[13251] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) is leaving simple_bridge technology [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Announcer/ARI-0000001b;2: [Aug 18 10:33:59] DEBUG[13672] channel.c: Channel Announcer/ARI-0000001b;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13677] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Announcer/ARI-0000001b;2: [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Announcer/ARI-0000001b;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is joining softmix technology [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: SIP/zvonobot-00000020: [Aug 18 10:33:59] DEBUG[13674] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13678] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is joining [Aug 18 10:33:59] DEBUG[13677] http.c: HTTP Request URI is /ari/channels/212986 [Aug 18 10:33:59] DEBUG[13642] stasis.c: Creating topic. name: channel:1629282839.182, detail: [Aug 18 10:33:59] DEBUG[13642] stasis.c: Topic 'channel:1629282839.182': 0x7f0ca00311d0 created [Aug 18 10:33:59] DEBUG[13642] stasis.c: Creating topic. name: cache:215/channel:1629282839.182, detail: [Aug 18 10:33:59] DEBUG[13642] stasis.c: Topic 'cache:215/channel:1629282839.182': 0x7f0ca0022980 created [Aug 18 10:33:59] DEBUG[13677] http.c: match request [ari/channels/212986] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13668] stasis.c: Creating topic. name: channel:1629282839.181, detail: [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13677] http.c: match request [ari/channels/212986] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13251] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13251] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13251] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212986-00000002 - start 1629282831.300515 answer 1629282831.300515 end 1629282839.607313 dur 8.306 bill 8.306 dispo ANSWERED [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13251] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13251] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: SIP/zvonobot-00000020: [Aug 18 10:33:59] DEBUG[13251] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824 is already using the new technology. [Aug 18 10:33:59] DEBUG[13677] http.c: match request [ari/channels/212986] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: SIP/zvonobot-00000020: Not in SFU mode [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13668] stasis.c: Topic 'channel:1629282839.181': 0x7f0c2c07fd10 created [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Recorder/ARI-00000011;2: [Aug 18 10:33:59] DEBUG[13668] stasis.c: Creating topic. name: cache:216/channel:1629282839.181, detail: [Aug 18 10:33:59] DEBUG[13672] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13677] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Finding handler for channels/212986 [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Finding handler for 212986 [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking channels create: Didn't match 212986 [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13677] res_ari.c: Checking channels externalMedia: Didn't match 212986 [Aug 18 10:33:59] DEBUG[13677] res_ari.c: No explicit handler found for 212986. Using wildcard channelId. [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13672] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13504] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: stopping mixing thread [Aug 18 10:33:59] DEBUG[13462] channel.c: Recorder/ARI-00000013;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:33:59] DEBUG[13671] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology destructor (deferred, dummy) [Aug 18 10:33:59] DEBUG[20534] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: Waiting for mixing thread to die. [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13668] stasis.c: Topic 'cache:216/channel:1629282839.181': 0x7f0c2c008a60 created [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Recorder/ARI-00000011;2: [Aug 18 10:33:59] DEBUG[13672] bridge_softmix.c: Recorder/ARI-00000011;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology start [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel 0x7f0c180b93f0 'Announcer/ARI-00000017;2' hanging up. Refs: 2 [Aug 18 10:33:59] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology destructor [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13674] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] VERBOSE[13671] bridge_channel.c: Channel Announcer/ARI-0000001d;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:59] DEBUG[13674] stasis.c: Creating topic. name: bridge:0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3, detail: [Aug 18 10:33:59] DEBUG[13454] channel.c: SIP/zvonobot-00000023: Dropping redundant connected line update "" <>. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13251] stasis/control.c: robot_212986, 8b092052-108a-4921-8aad-1aecb4e2c824: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[13674] stasis.c: Topic 'bridge:0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3': 0x7f0c3c023950 created [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[13251] stasis/app.c: bridge '8b092052-108a-4921-8aad-1aecb4e2c824': is 3 interested in calls_0 [Aug 18 10:33:59] DEBUG[13248] stasis/control.c: robot_212986: Channel departing bridge [Aug 18 10:33:59] DEBUG[13674] stasis.c: Creating topic. name: cache:217/bridge:0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3, detail: [Aug 18 10:33:59] DEBUG[13248] bridge.c: Waiting for 0x7f0cb402e7f0(UnicastRTP/127.0.0.1:50291-0x7f0ca0023720) bridge thread to die. [Aug 18 10:33:59] DEBUG[20535] devicestate.c: Changing state for Recorder/ARI - state 2 (In use) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:33:59] DEBUG[13637] stasis.c: Creating topic. name: channel:1629282839.183, detail: [Aug 18 10:33:59] DEBUG[13635] channel.c: Channel 0x7f0c9406b7d0 'Recorder/ARI-0000001e;2' allocated [Aug 18 10:33:59] DEBUG[13635] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] DEBUG[13637] stasis.c: Topic 'channel:1629282839.183': 0x7f0ca80574d0 created [Aug 18 10:33:59] DEBUG[13637] stasis.c: Creating topic. name: cache:218/channel:1629282839.183, detail: [Aug 18 10:33:59] DEBUG[13637] stasis.c: Topic 'cache:218/channel:1629282839.183': 0x7f0ca8009410 created [Aug 18 10:33:59] DEBUG[13251] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[13248] stasis/app.c: channel 'robot_212986': is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13248] channel.c: Channel 0x7f0ca002fd30 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720' hanging up. Refs: 2 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:33:59] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:33:59] DEBUG[13674] stasis.c: Topic 'cache:217/bridge:0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3': 0x7f0c3c069eb0 created [Aug 18 10:33:59] DEBUG[20616] app_queue.c: Device 'Recorder/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13674] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #64 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #64)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13674] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13674] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13674] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:59] DEBUG[13674] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13674] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[13674] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13671] bridge.c: Chose bridge technology softmix [Aug 18 10:33:59] VERBOSE[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: switching from simple_bridge technology to softmix [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology constructor [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c9006b170(SIP/zvonobot-0000002e) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c8808e340(Recorder/ARI-00000014;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13674] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology stop [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9c048790(Announcer/ARI-0000001d;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Announcer/ARI-0000001d;2: [Aug 18 10:33:59] DEBUG[13676] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: starting mixing thread [Aug 18 10:33:59] DEBUG[13674] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13671] channel.c: Channel Announcer/ARI-0000001d;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Announcer/ARI-0000001d;2: [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Announcer/ARI-0000001d;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is joining softmix technology [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: SIP/zvonobot-0000002e: [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:33:59] DEBUG[13679] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is joining [Aug 18 10:33:59] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP audio difference is 12968, ms is 1641 [Aug 18 10:33:59] DEBUG[13557] res_stasis_playback.c: 1629282838.161: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 From: ;tag=as574e1b12 To: ;tag=as2c28f3a7 Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 595882522 595882522 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18112 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as574e1b12 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2c28f3a7 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 595882522 595882522 IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18112 RTP/AVP 0 8 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 (Checking To) --From tag as574e1b12 --To-tag as2c28f3a7 [Aug 18 10:33:59] DEBUG[13557] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:59] DEBUG[13682] channel.c: Channel Announcer/ARI-0000001b;1 setting write format path: gsm -> slin [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: SIP/zvonobot-0000002e: [Aug 18 10:33:59] DEBUG[13557] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: SIP/zvonobot-0000002e: Not in SFU mode [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Recorder/ARI-00000014;2: [Aug 18 10:33:59] DEBUG[13671] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13671] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Recorder/ARI-00000014;2: [Aug 18 10:33:59] DEBUG[13671] bridge_softmix.c: Recorder/ARI-00000014;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology start [Aug 18 10:33:59] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology destructor [Aug 18 10:33:59] DEBUG[13680] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: starting mixing thread [Aug 18 10:33:59] DEBUG[13682] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:59] VERBOSE[13682] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:59] DEBUG[13573] res_stasis_playback.c: 1629282838.171: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:33:59] DEBUG[13573] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:33:59] DEBUG[13681] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13573] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:33:59] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP audio difference is 2960, ms is 390 [Aug 18 10:33:59] DEBUG[13681] http.c: HTTP Request URI is /ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/addChannel?channel=213009 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Got SDP version 595882522 and unique parts [root 595882522 IN IP4 178.62.121.41] [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 595882522 595882522 IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:33:59] DEBUG[13679] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pushing 0x7f0c94055bb0(Recorder/ARI-0000001e;2) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[13681] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13681] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) ICE set role failed; no ice instance [Aug 18 10:33:59] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:33:59] DEBUG[13681] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/addChannel] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13678] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pushing 0x7f0c78074930(Recorder/ARI-0000001c;2) [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP setting address on RTP instance [Aug 18 10:33:59] DEBUG[13683] channel.c: Channel Announcer/ARI-0000001d;1 setting write format path: gsm -> slin [Aug 18 10:33:59] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c1c022950 -- Strict RTP learning after remote address set to: 178.62.121.41:18112 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:18112 [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0078108) from 0x7f0c147e2330 to 0x7f0c1c00be18 [Aug 18 10:33:59] DEBUG[13683] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00645a8) from 0x7f0c147e2330 to 0x7f0c1c00be18 [Aug 18 10:33:59] DEBUG[13681] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00c4328) from 0x7f0c147e2330 to 0x7f0c1c00be18 [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP ignoring duplicate property [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000013 setting read format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000013 setting write format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) DTLS - ast_rtp_activate rtp=0x7f0c1c022950 - setup and perform DTLS' [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c022950) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c022950) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:33:59] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] VERBOSE[13683] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:33:59] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:33:59] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117058@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK177a3285 Max-Forwards: 70 From: ;tag=as574e1b12 To: ;tag=as2c28f3a7 Contact: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Finding handler for bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/addChannel [Aug 18 10:33:59] VERBOSE[12971] dial.c: SIP/zvonobot-00000013 answered [Aug 18 10:33:59] VERBOSE[12971] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000013 [Aug 18 10:33:59] DEBUG[12971] stasis/app.c: Channel '212982' is 2 interested in calls_0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (1) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[12971] res_rtp_asterisk.c: 0x7f0c1c022950 -- Strict RTP switching to RTP target address 178.62.121.41:18112 as source [Aug 18 10:33:59] DEBUG[12971] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:33:59] DEBUG[12971] channel.c: Channel SIP/zvonobot-00000013 setting read format path: ulaw -> alaw [Aug 18 10:33:59] DEBUG[12971] channel.c: Channel SIP/zvonobot-00000013 setting write format path: alaw -> ulaw [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Session timer started: 59 - 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 1768000ms [Aug 18 10:33:59] DEBUG[13679] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] VERBOSE[13679] bridge_channel.c: Channel Recorder/ARI-0000001e;2 joined 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:33:59] DEBUG[13684] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13684] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:59] DEBUG[13684] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13684] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13684] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13684] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13649] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 From: ;tag=as30c9fc1b To: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag [Aug 18 10:33:59] DEBUG[13611] channel.c: Channel 0x7f0c240f5d70 'Announcer/ARI-0000001f;1' allocated [Aug 18 10:33:59] DEBUG[13611] stasis.c: Creating topic. name: channel:1629282839.184, detail: [Aug 18 10:33:59] DEBUG[13611] stasis.c: Topic 'channel:1629282839.184': 0x7f0c24049eb0 created [Aug 18 10:33:59] DEBUG[13611] stasis.c: Creating topic. name: cache:219/channel:1629282839.184, detail: [Aug 18 10:33:59] DEBUG[13611] stasis.c: Topic 'cache:219/channel:1629282839.184': 0x7f0c240f8cc0 created [Aug 18 10:33:59] DEBUG[13649] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13678] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] DEBUG[13498] channel.c: Channel 0x7f0c180b93f0 'Announcer/ARI-00000017;2' destroying [Aug 18 10:33:59] DEBUG[13642] channel.c: Channel 0x7f0ca0076bd0 'Snoop/213008-0000000a' allocated [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13668] channel.c: Channel 0x7f0c2c08ce90 'Recorder/ARI-00000020;1' allocated [Aug 18 10:33:59] DEBUG[13685] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13642] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13642] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13668] stasis.c: Creating topic. name: channel:1629282839.185, detail: [Aug 18 10:33:59] DEBUG[13668] stasis.c: Topic 'channel:1629282839.185': 0x7f0c2c0f38f0 created [Aug 18 10:33:59] DEBUG[13668] stasis.c: Creating topic. name: cache:220/channel:1629282839.185, detail: [Aug 18 10:33:59] DEBUG[13668] stasis.c: Topic 'cache:220/channel:1629282839.185': 0x7f0c2c0f3ac0 created [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13679] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee'. Checking compatability for channels 'SIP/zvonobot-0000003b' and 'Recorder/ARI-0000001e;2' [Aug 18 10:33:59] DEBUG[13556] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060'. Checking compatability for channels 'SIP/zvonobot-0000002b' and 'Recorder/ARI-0000001a;2' [Aug 18 10:33:59] VERBOSE[13678] bridge_channel.c: Channel Recorder/ARI-0000001c;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:59] DEBUG[13694] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13556] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as channel 'SIP/zvonobot-0000002b' has features which prevent it [Aug 18 10:33:59] DEBUG[13684] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13556] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13556] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13694] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213008&app=calls_0&format=slin16&external_host=127.0.0.1%3A50118 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Finding handler for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13681] res_ari.c: No explicit handler found for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Finding handler for addChannel [Aug 18 10:33:59] DEBUG[13681] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:59] DEBUG[13681] stasis/control.c: 213009: Sending channel add_to_bridge command [Aug 18 10:33:59] DEBUG[13556] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13679] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as could not get details [Aug 18 10:33:59] DEBUG[13679] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13679] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13679] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13679] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13679] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee is already using the new technology. [Aug 18 10:33:59] DEBUG[13679] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel Recorder/ARI-0000001e;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel SIP/zvonobot-0000003b setting write format path: slin -> ulaw [Aug 18 10:33:59] DEBUG[13198] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000002f [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[13694] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13685] http.c: HTTP Request URI is /ari/channels/1629282832.85 [Aug 18 10:33:59] DEBUG[13198] stasis/control.c: 213009: Adding to bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 [Aug 18 10:33:59] DEBUG[13694] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13691] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13556] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13198] stasis/app.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13556] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 is already using the new technology. [Aug 18 10:33:59] DEBUG[13498] stasis.c: Destroying topic. name: cache:167/channel:1629282835.140, detail: [Aug 18 10:33:59] DEBUG[13498] stasis.c: Topic 'cache:167/channel:1629282835.140': 0x7f0c180bac50 destroyed [Aug 18 10:33:59] DEBUG[13498] stasis.c: Destroying topic. name: channel:1629282835.140, detail: [Aug 18 10:33:59] DEBUG[13684] stasis.c: Creating topic. name: bridge:45640e14-e267-477d-81ea-fbac374f9677, detail: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[13684] stasis.c: Topic 'bridge:45640e14-e267-477d-81ea-fbac374f9677': 0x7f0c8c03d010 created [Aug 18 10:33:59] DEBUG[13684] stasis.c: Creating topic. name: cache:221/bridge:45640e14-e267-477d-81ea-fbac374f9677, detail: [Aug 18 10:33:59] DEBUG[13684] stasis.c: Topic 'cache:221/bridge:45640e14-e267-477d-81ea-fbac374f9677': 0x7f0c8c058830 created [Aug 18 10:33:59] DEBUG[13684] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13684] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13684] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13684] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:59] DEBUG[13684] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13684] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[13684] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel SIP/zvonobot-0000003b setting read format path: ulaw -> slin [Aug 18 10:33:59] DEBUG[13498] stasis.c: Topic 'channel:1629282835.140': 0x7f0c18090c60 destroyed [Aug 18 10:33:59] DEBUG[13694] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13695] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is joining [Aug 18 10:33:59] DEBUG[13685] http.c: match request [ari/channels/1629282832.85] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13691] http.c: HTTP Request URI is /ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play?media=sound%3Asilence%2F2 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[13694] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #64 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #64)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7;received=159.65.48.104 From: ;tag=as0e0b214d To: ;tag=as285b992f Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dbc5b08" Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e165ff7;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as285b992f [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dbc5b08" [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag as285b992f [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 From: ;tag=as2b432b30 To: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13684] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #37 - INVITE (got response) [Aug 18 10:33:59] DEBUG[13684] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13691] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13635] res_stasis_recording.c: 1629282838.174: Sending record(213023_xsTqdZZuLTcscXpKpHHMfkbIildpHaqk.wav) command [Aug 18 10:33:59] DEBUG[13685] http.c: match request [ari/channels/1629282832.85] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[13685] http.c: match request [ari/channels/1629282832.85] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13691] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13691] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13635] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13678] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52'. Checking compatability for channels 'SIP/zvonobot-00000030' and 'Recorder/ARI-0000001c;2' [Aug 18 10:33:59] DEBUG[13635] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13685] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13691] http.c: Match made with [ari] [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 500 Server error Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 From: ;tag=as2b432b30 To: ;tag=as329ffcb0 Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Retry-After: 7 Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 500 Server error [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2dcb18be;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as329ffcb0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 14]: Retry-After: 7 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag as329ffcb0 [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Finding handler for bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play [Aug 18 10:33:59] DEBUG[13697] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Finding handler for channels/1629282832.85 [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[13699] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13697] http.c: HTTP Request URI is /ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/addChannel?channel=212982 [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Stopping retransmission on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Finding handler for externalMedia [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel Recorder/ARI-0000001e;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:33:59] DEBUG[13679] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: alaw -> slin [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13697] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/addChannel] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13699] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:33:59] DEBUG[13695] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: pushing 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) [Aug 18 10:33:59] DEBUG[13678] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as could not get details [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/addChannel] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13694] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117001@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f Max-Forwards: 70 From: ;tag=as2b432b30 To: Contact: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13698] app.c: play_and_record: , /var/spool/asterisk/recording/213023_xsTqdZZuLTcscXpKpHHMfkbIildpHaqk, 'wav' [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/addChannel] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13677] channel.c: Soft-Hanging (0x20) up channel 'SIP/zvonobot-00000016' [Aug 18 10:33:59] DEBUG[13697] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13678] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13694] netsock2.c: Splitting '127.0.0.1:50118' into... [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[13699] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13248] res_rtp_asterisk.c: (0x7f0ca0023720) DTLS stop [Aug 18 10:33:59] DEBUG[13248] res_rtp_asterisk.c: (0x7f0ca0023720) DTLS srtp - stopped timeout timer' [Aug 18 10:33:59] DEBUG[13248] res_rtp_asterisk.c: (0x7f0ca0023720) ICE RTP transport deallocating [Aug 18 10:33:59] DEBUG[13248] res_rtp_asterisk.c: (0x7f0ca0023720) ICE stopped [Aug 18 10:33:59] DEBUG[13248] rtp_engine.c: Destroyed RTP instance '0x7f0ca0023720' [Aug 18 10:33:59] DEBUG[13248] channel.c: Channel 0x7f0ca002fd30 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720' destroying [Aug 18 10:33:59] DEBUG[13677] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13678] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (2) INVITE - 5 [Aug 18 10:33:59] DEBUG[13677] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:33:59] DEBUG[13698] app.c: Recording Formats: sfmts=wav [Aug 18 10:33:59] VERBOSE[13695] bridge_channel.c: Channel SIP/zvonobot-0000002f joined 'simple_bridge' stasis-bridge <0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3> [Aug 18 10:33:59] DEBUG[13686] stasis/app.c: Channel '1629282839.182' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13668] channel.c: Channel 0x7f0c2c096fd0 'Recorder/ARI-00000020;2' allocated [Aug 18 10:33:59] DEBUG[13668] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] DEBUG[13694] netsock2.c: ...host '127.0.0.1' and port '50118'. [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Finding handler for 1629282832.85 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13611] channel.c: Channel 0x7f0c24016f20 'Announcer/ARI-0000001f;2' allocated [Aug 18 10:33:59] DEBUG[13611] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] DEBUG[13611] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001f;1' [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Finding handler for bridges/45640e14-e267-477d-81ea-fbac374f9677/addChannel [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Finding handler for 45640e14-e267-477d-81ea-fbac374f9677 [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13697] res_ari.c: No explicit handler found for 45640e14-e267-477d-81ea-fbac374f9677. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Finding handler for addChannel [Aug 18 10:33:59] DEBUG[13697] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:33:59] DEBUG[13697] stasis/control.c: 212982: Sending channel add_to_bridge command [Aug 18 10:33:59] DEBUG[13699] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13699] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13701] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13694] netsock2.c: Splitting '127.0.0.1:50118' into... [Aug 18 10:33:59] DEBUG[13694] netsock2.c: ...host '127.0.0.1' and port '50118'. [Aug 18 10:33:59] DEBUG[13694] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:33:59] DEBUG[13694] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca804bf40' [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP allocated port 11868 [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE creating session 127.0.0.1:11868 (11868) [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE create [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE add system candidates [Aug 18 10:33:59] DEBUG[13694] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:33:59] DEBUG[13694] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE add candidate: 159.65.48.104:11868, 2130706431 [Aug 18 10:33:59] DEBUG[13694] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:33:59] DEBUG[13694] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:33:59] DEBUG[13694] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE add candidate: 10.131.0.10:11868, 2130706431 [Aug 18 10:33:59] DEBUG[13694] rtp_engine.c: RTP instance '0x7f0ca804bf40' is setup and ready to go [Aug 18 10:33:59] DEBUG[13694] stasis.c: Creating topic. name: channel:robot_213008, detail: [Aug 18 10:33:59] DEBUG[13694] stasis.c: Topic 'channel:robot_213008': 0x7f0ca806d770 created [Aug 18 10:33:59] DEBUG[13694] stasis.c: Creating topic. name: cache:222/channel:robot_213008, detail: [Aug 18 10:33:59] DEBUG[13694] stasis.c: Topic 'cache:222/channel:robot_213008': 0x7f0ca806e1b0 created [Aug 18 10:33:59] DEBUG[13701] http.c: HTTP Request URI is /ari/channels/1629282831.69 [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking channels create: Didn't match 1629282832.85 [Aug 18 10:33:59] DEBUG[13702] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is joining [Aug 18 10:33:59] DEBUG[13703] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c240f8830(Announcer/ARI-0000001f;2) is joining [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel 'robot_212986': is 0 interested in calls_0 [Aug 18 10:33:59] DEBUG[20620] stasis/app.c: channel 'robot_212986' unsubscribed from calls_0 [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: cache:85/channel:robot_212986, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'cache:85/channel:robot_212986': 0x7f0ca002e2f0 destroyed [Aug 18 10:33:59] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212986, detail: [Aug 18 10:33:59] DEBUG[20620] stasis.c: Topic 'channel:robot_212986': 0x7f0ca002f0e0 destroyed [Aug 18 10:33:59] DEBUG[13701] http.c: match request [ari/channels/1629282831.69] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13678] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13701] http.c: match request [ari/channels/1629282831.69] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13701] http.c: match request [ari/channels/1629282831.69] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13701] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Finding handler for channels/1629282831.69 [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Finding handler for 1629282831.69 [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking channels create: Didn't match 1629282831.69 [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13701] res_ari.c: Checking channels externalMedia: Didn't match 1629282831.69 [Aug 18 10:33:59] DEBUG[13701] res_ari.c: No explicit handler found for 1629282831.69. Using wildcard channelId. [Aug 18 10:33:59] DEBUG[12971] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000013 [Aug 18 10:33:59] DEBUG[12971] stasis/control.c: 212982: Adding to bridge 45640e14-e267-477d-81ea-fbac374f9677 [Aug 18 10:33:59] DEBUG[12971] stasis/app.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Recorder - ARI [Aug 18 10:33:59] DEBUG[13637] channel.c: Channel 0x7f0ca800d0a0 'Snoop/212977-0000000b' allocated [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 From: ;tag=as3edf3f1c To: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag [Aug 18 10:33:59] DEBUG[13678] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Setting 0x7f0c30064e70(SIP/zvonobot-00000016) state from:0 to:1 [Aug 18 10:33:59] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282839.187, detail: [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13550] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6'. Checking compatability for channels 'SIP/zvonobot-0000000e' and 'Recorder/ARI-00000019;2' [Aug 18 10:33:59] DEBUG[13550] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as channel 'SIP/zvonobot-0000000e' has features which prevent it [Aug 18 10:33:59] DEBUG[13550] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13550] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13550] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13550] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 is already using the new technology. [Aug 18 10:33:59] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 is already using the new technology. [Aug 18 10:33:59] DEBUG[13702] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pushing 0x7f0c2c08b700(Recorder/ARI-00000020;2) [Aug 18 10:33:59] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13637] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13637] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Finding handler for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13691] res_ari.c: No explicit handler found for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Finding handler for play [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:59] DEBUG[13691] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:59] DEBUG[13691] stasis.c: Creating topic. name: channel:1629282839.188, detail: [Aug 18 10:33:59] DEBUG[13695] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13695] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13695] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13695] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13695] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13695] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 is already using the new technology. [Aug 18 10:33:59] DEBUG[13695] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is joining simple_bridge technology [Aug 18 10:33:59] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:33:59] DEBUG[13686] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 14 instead [Aug 18 10:33:59] VERBOSE[13698] app.c: x=0, open writing: /var/spool/asterisk/recording/213023_xsTqdZZuLTcscXpKpHHMfkbIildpHaqk format: wav, 0x7f0ca0022f20 [Aug 18 10:33:59] DEBUG[13691] stasis.c: Topic 'channel:1629282839.188': 0x7f0c90059200 created [Aug 18 10:33:59] DEBUG[13691] stasis.c: Creating topic. name: cache:223/channel:1629282839.188, detail: [Aug 18 10:33:59] DEBUG[13691] stasis.c: Topic 'cache:223/channel:1629282839.188': 0x7f0c9005c460 created [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13699] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13699] stasis.c: Creating topic. name: bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc, detail: [Aug 18 10:33:59] DEBUG[13699] stasis.c: Topic 'bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc': 0x7f0ca40044e0 created [Aug 18 10:33:59] DEBUG[13699] stasis.c: Creating topic. name: cache:224/bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc, detail: [Aug 18 10:33:59] DEBUG[13699] stasis.c: Topic 'cache:224/bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc': 0x7f0ca402b040 created [Aug 18 10:33:59] DEBUG[13699] bridge_native_rtp.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13699] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13699] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13699] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:33:59] DEBUG[13699] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13699] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: calling simple_bridge technology constructor [Aug 18 10:33:59] DEBUG[13699] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: calling simple_bridge technology start [Aug 18 10:33:59] DEBUG[13699] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:33:59] DEBUG[13703] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pushing 0x7f0c240f8830(Announcer/ARI-0000001f;2) [Aug 18 10:33:59] DEBUG[13703] bridge_roles.c: Set role 'announcer' [Aug 18 10:33:59] VERBOSE[13703] bridge_channel.c: Channel Announcer/ARI-0000001f;2 joined 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13703] bridge.c: Chose bridge technology softmix [Aug 18 10:33:59] VERBOSE[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: switching from simple_bridge technology to softmix [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology constructor [Aug 18 10:33:59] DEBUG[13685] res_ari.c: Checking channels externalMedia: Didn't match 1629282832.85 [Aug 18 10:33:59] DEBUG[13685] res_ari.c: No explicit handler found for 1629282832.85. Using wildcard channelId. [Aug 18 10:33:59] DEBUG[13716] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.187': 0x7f0c30079200 created [Aug 18 10:33:59] DEBUG[20545] stasis.c: Creating topic. name: cache:225/channel:1629282839.187, detail: [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #33 - INVITE (got response) [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' Request 103: Found [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:33:59] DEBUG[13699] http.c: HTTP closing session. Top level [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0ca0053060(SIP/zvonobot-00000023) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0c7c018d60(Recorder/ARI-00000013;2) to dummy bridge temporarily [Aug 18 10:33:59] DEBUG[13711] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:33:59] DEBUG[13702] bridge_roles.c: Set role 'recorder' [Aug 18 10:33:59] VERBOSE[13702] bridge_channel.c: Channel Recorder/ARI-00000020;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:225/channel:1629282839.187': 0x7f0c300924c0 created [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13678] channel.c: Channel Recorder/ARI-0000001c;2 setting read format path: slin -> slin [Aug 18 10:33:59] DEBUG[13704] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining [Aug 18 10:33:59] DEBUG[13711] http.c: HTTP Request URI is /ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play?media=sound%3Asilence%2F2 [Aug 18 10:33:59] DEBUG[13711] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13711] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13711] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13711] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Finding handler for bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Finding handler for bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Finding handler for d0f9af3e-7f00-4d11-8990-3d67ba7213d6 [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:33:59] DEBUG[13711] res_ari.c: No explicit handler found for d0f9af3e-7f00-4d11-8990-3d67ba7213d6. Using wildcard bridgeId. [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Finding handler for play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:33:59] DEBUG[13711] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:33:59] DEBUG[13711] stasis.c: Creating topic. name: channel:1629282839.189, detail: [Aug 18 10:33:59] DEBUG[13711] stasis.c: Topic 'channel:1629282839.189': 0x7f0c0807fe00 created [Aug 18 10:33:59] DEBUG[13711] stasis.c: Creating topic. name: cache:226/channel:1629282839.189, detail: [Aug 18 10:33:59] DEBUG[13711] stasis.c: Topic 'cache:226/channel:1629282839.189': 0x7f0c0805c660 created [Aug 18 10:33:59] DEBUG[13714] http.c: HTTP opening session. Top level [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pulling 0x7f0c30064e70(SIP/zvonobot-00000016) [Aug 18 10:33:59] VERBOSE[13232] bridge_channel.c: Channel SIP/zvonobot-00000016 left 'simple_bridge' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c30064e70(SIP/zvonobot-00000016) is leaving simple_bridge technology [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Setting 0x7f0c8c037dd0(Recorder/ARI-00000005;2) state from:0 to:2 [Aug 18 10:33:59] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13716] http.c: HTTP Request URI is /ari/channels/213023/snoop?app=calls_0&spy=in [Aug 18 10:33:59] DEBUG[13716] http.c: match request [ari/channels/213023/snoop] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13716] http.c: match request [ari/channels/213023/snoop] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13716] http.c: match request [ari/channels/213023/snoop] with handler [ari] len 3 [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is leaving simple_bridge technology (dummy) [Aug 18 10:33:59] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: cache:225/channel:1629282839.187, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'cache:225/channel:1629282839.187': 0x7f0c300924c0 destroyed [Aug 18 10:33:59] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282839.187, detail: [Aug 18 10:33:59] DEBUG[20545] stasis.c: Topic 'channel:1629282839.187': 0x7f0c30079200 destroyed [Aug 18 10:33:59] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:51', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212986', '')] [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (4) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117006@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff Max-Forwards: 70 From: ;tag=as3edf3f1c To: Contact: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117006@178.62.121.41", nonce="2e114138", response="546d7b2d9fa7f2afdf06a7c01e9051cf" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1806379122 1806379123 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15562 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (3) INVITE - 5 [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:33:59] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:33:59] DEBUG[13716] http.c: Match made with [ari] [Aug 18 10:33:59] DEBUG[13678] channel.c: Channel SIP/zvonobot-00000030 setting write format path: slin -> ulaw [Aug 18 10:33:59] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:33:59] DEBUG[13686] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13232] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13232] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13232] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13232] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13232] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13232] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 is already using the new technology. [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Bridge is returning 0x7f0c30064e70(SIP/zvonobot-00000016) to read format alaw [Aug 18 10:33:59] DEBUG[13232] channel.c: Channel SIP/zvonobot-00000016 setting read format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[13232] bridge_channel.c: Bridge is returning 0x7f0c30064e70(SIP/zvonobot-00000016) to write format alaw [Aug 18 10:33:59] DEBUG[13232] channel.c: Channel SIP/zvonobot-00000016 setting write format path: alaw -> alaw [Aug 18 10:33:59] DEBUG[13232] stasis/control.c: 212986, 90245cdf-0ee9-4414-b99e-c22349f119a2: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[13232] stasis/app.c: bridge '90245cdf-0ee9-4414-b99e-c22349f119a2': is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13232] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:33:59] DEBUG[12991] stasis/control.c: 212986: Channel departing bridge [Aug 18 10:33:59] DEBUG[12991] bridge.c: Waiting for 0x7f0c30064e70(SIP/zvonobot-00000016) bridge thread to die. [Aug 18 10:33:59] DEBUG[12991] stasis/app.c: channel '212986': is 1 interested in calls_0 [Aug 18 10:33:59] DEBUG[13234] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: pulling 0x7f0c8c037dd0(Recorder/ARI-00000005;2) [Aug 18 10:33:59] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000016 - start 1629282826.058560 answer 1629282831.078152 end 1629282839.981569 dur 13.923 bill 8.903 dispo ANSWERED [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology stop [Aug 18 10:33:59] VERBOSE[13234] bridge_channel.c: Channel Recorder/ARI-00000005;2 left 'simple_bridge' stasis-bridge <90245cdf-0ee9-4414-b99e-c22349f119a2> [Aug 18 10:33:59] DEBUG[13716] res_ari.c: Finding handler for channels/213023/snoop [Aug 18 10:33:59] DEBUG[13714] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212977&app=calls_0&format=slin16&external_host=127.0.0.1%3A50194 [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c240f8830(Announcer/ARI-0000001f;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13234] bridge_channel.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: 0x7f0c8c037dd0(Recorder/ARI-00000005;2) is leaving simple_bridge technology [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Announcer/ARI-0000001f;2: [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:33:59] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:33:59] DEBUG[12991] channel.c: Channel 0x7f0c34024da0 'SIP/zvonobot-00000016' hanging up. Refs: 3 [Aug 18 10:33:59] DEBUG[13234] bridge_native_rtp.c: Bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' can not use native RTP bridge as two channels are required [Aug 18 10:33:59] DEBUG[13234] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13234] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13234] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13234] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:33:59] DEBUG[13234] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2 is already using the new technology. [Aug 18 10:33:59] DEBUG[13234] channel.c: Channel 0x7f0c8c037210 'Recorder/ARI-00000005;2' hanging up. Refs: 2 [Aug 18 10:33:59] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:33:59] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP changing ssrc from 501991794 to 124785352 due to a source change [Aug 18 10:33:59] DEBUG[13703] channel.c: Channel Announcer/ARI-0000001f;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13714] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:33:59] DEBUG[13702] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d'. Checking compatability for channels 'SIP/zvonobot-0000002a' and 'Recorder/ARI-00000020;2' [Aug 18 10:33:59] DEBUG[13716] res_ari.c: Finding handler for channels [Aug 18 10:33:59] DEBUG[13198] stasis/app.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' is 2 interested in calls_0 [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:33:59] DEBUG[13702] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as could not get details [Aug 18 10:33:59] DEBUG[13702] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:33:59] DEBUG[13702] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Announcer/ARI-0000001f;2: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Announcer/ARI-0000001f;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining softmix technology [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: SIP/zvonobot-00000023: [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: SIP/zvonobot-00000023: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: SIP/zvonobot-00000023: Not in SFU mode [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining softmix technology [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Recorder/ARI-00000013;2: [Aug 18 10:33:59] DEBUG[13703] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:33:59] DEBUG[13703] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Recorder/ARI-00000013;2: [Aug 18 10:33:59] DEBUG[13703] bridge_softmix.c: Recorder/ARI-00000013;2: Not in SFU mode [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology start [Aug 18 10:33:59] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13706] stasis/app.c: Channel '1629282839.183' is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13706] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 17 instead [Aug 18 10:33:59] DEBUG[13714] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:33:59] DEBUG[13681] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13681] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13702] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13702] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13702] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13702] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13714] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:00] DEBUG[13714] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13717] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13720] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Finding handler for 213023 [Aug 18 10:34:00] DEBUG[13720] http.c: HTTP Request URI is /ari/channels/213056?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116984&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channels create: Didn't match 213023 [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channels externalMedia: Didn't match 213023 [Aug 18 10:34:00] DEBUG[13716] res_ari.c: No explicit handler found for 213023. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Finding handler for snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:00] DEBUG[13716] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:00] DEBUG[13717] http.c: HTTP Request URI is /ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/record?name=213009_KnHWGInDjuQFpdGbHGchCssGOUDQXoGO&format=wav [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel Recorder/ARI-00000020;2 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 736, ms is 66 [Aug 18 10:34:00] DEBUG[13717] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/record] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13720] http.c: match request [ari/channels/213056] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13704] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pushing 0x7f0c18091350(SIP/zvonobot-00000013) [Aug 18 10:34:00] VERBOSE[13056] res_rtp_asterisk.c: 0x7f0cb0010680 -- Strict RTP learning complete - Locking on source address 178.62.121.41:11670 [Aug 18 10:34:00] DEBUG[13717] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/record] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13717] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/record] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13717] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Finding handler for bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Finding handler for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13717] res_ari.c: No explicit handler found for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Finding handler for record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:00] DEBUG[13717] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:00] DEBUG[13717] stasis.c: Creating topic. name: channel:1629282840.190, detail: [Aug 18 10:34:00] DEBUG[13717] stasis.c: Topic 'channel:1629282840.190': 0x7f0c2008c040 created [Aug 18 10:34:00] DEBUG[13717] stasis.c: Creating topic. name: cache:227/channel:1629282840.190, detail: [Aug 18 10:34:00] DEBUG[13717] stasis.c: Topic 'cache:227/channel:1629282840.190': 0x7f0c2008ca40 created [Aug 18 10:34:00] DEBUG[13723] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13723] http.c: HTTP Request URI is /ari/channels/213054?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116986&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13723] http.c: match request [ari/channels/213054] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13723] http.c: match request [ari/channels/213054] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13723] http.c: match request [ari/channels/213054] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13723] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13723] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Finding handler for channels/213054 [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Finding handler for 213054 [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking channels create: Didn't match 213054 [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13723] res_ari.c: Checking channels externalMedia: Didn't match 213054 [Aug 18 10:34:00] DEBUG[13056] bridge_softmix.c: Frame type 10 unsupported [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[13720] http.c: match request [ari/channels/213056] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13720] http.c: match request [ari/channels/213056] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13720] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13720] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Finding handler for channels/213056 [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Finding handler for 213056 [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking channels create: Didn't match 213056 [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13720] res_ari.c: Checking channels externalMedia: Didn't match 213056 [Aug 18 10:34:00] DEBUG[13720] res_ari.c: No explicit handler found for 213056. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel SIP/zvonobot-0000002a setting write format path: slin -> ulaw [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel SIP/zvonobot-0000002a setting read format path: ulaw -> slin [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel Recorder/ARI-00000020;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Finding handler for channels [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 Max-Forwards: 70 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6be15af9 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0453a0d2 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking From) --From tag as6be15af9 --To-tag as0453a0d2 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:00] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:00] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13723] res_ari.c: No explicit handler found for 213054. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13715] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: starting mixing thread [Aug 18 10:34:00] DEBUG[13678] channel.c: Channel SIP/zvonobot-00000030 setting read format path: ulaw -> slin [Aug 18 10:34:00] VERBOSE[13704] bridge_channel.c: Channel SIP/zvonobot-00000013 joined 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:00] DEBUG[13611] res_stasis_playback.c: 1629282839.179: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:00] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13611] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:00] DEBUG[13732] channel.c: Channel Announcer/ARI-0000001f;1 setting write format path: gsm -> slin [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13611] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13732] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:00] VERBOSE[13732] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 944, ms is 79 [Aug 18 10:34:00] DEBUG[13736] http.c: HTTP opening session. Top level [Aug 18 10:34:00] VERBOSE[13441] res_rtp_asterisk.c: 0x7f0c8c013990 -- Strict RTP learning complete - Locking on source address 178.62.121.41:10912 [Aug 18 10:34:00] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 944, ms is 138 [Aug 18 10:34:00] DEBUG[13728] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13736] http.c: HTTP Request URI is /ari/channels/213057?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116983&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13678] channel.c: Channel Recorder/ARI-0000001c;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: Allocating new SIP dialog for 1413cf3f4a3226574abfed386569ce53@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13720] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c08f640' [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) RTP allocated port 19866 [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE creating session 0.0.0.0:19866 (19866) [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE create [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:00] DEBUG[13694] channel.c: Channel 0x7f0ca806b9f0 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40' allocated [Aug 18 10:34:00] DEBUG[13694] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:00] VERBOSE[13694] res_rtp_asterisk.c: 0x7f0ca8066bc0 -- Strict RTP learning after remote address set to: 127.0.0.1:50118 [Aug 18 10:34:00] DEBUG[13668] res_stasis_recording.c: 1629282839.181: Sending record(213007_pPaTJtaVJwuRGbJSRCzOzmloUbPCTlFs.wav) command [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13694] res_stasis.c: calls_0: Subscribing to robot_213008 [Aug 18 10:34:00] DEBUG[13694] stasis/app.c: Channel 'robot_213008' is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13694] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:00] DEBUG[13728] http.c: HTTP Request URI is /ari/channels/213055?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116985&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE add system candidates [Aug 18 10:34:00] DEBUG[13720] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13720] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE add candidate: 159.65.48.104:19866, 2130706431 [Aug 18 10:34:00] DEBUG[13738] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13694] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[20535] devicestate.c: Changing state for Recorder/ARI - state 2 (In use) [Aug 18 10:34:00] DEBUG[13250] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13668] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:00] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:00] DEBUG[13668] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13733] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[20616] app_queue.c: Device 'Recorder/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:00] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP audio difference is 704, ms is 108 [Aug 18 10:34:00] DEBUG[13691] channel.c: Channel 0x7f0c90063430 'Announcer/ARI-00000021;1' allocated [Aug 18 10:34:00] DEBUG[13691] stasis.c: Creating topic. name: channel:1629282840.191, detail: [Aug 18 10:34:00] DEBUG[13691] stasis.c: Topic 'channel:1629282840.191': 0x7f0c90062c40 created [Aug 18 10:34:00] DEBUG[13691] stasis.c: Creating topic. name: cache:228/channel:1629282840.191, detail: [Aug 18 10:34:00] DEBUG[13691] stasis.c: Topic 'cache:228/channel:1629282840.191': 0x7f0c9005c640 created [Aug 18 10:34:00] DEBUG[13738] http.c: HTTP Request URI is /ari/channels/213059?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116981&callerId=74950493843 [Aug 18 10:34:00] DEBUG[12991] chan_sip.c: Hangup call SIP/zvonobot-00000016, SIP callid 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[12991] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS srtp - stopped timeout timer' [Aug 18 10:34:00] DEBUG[12991] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS srtp - stopped timeout timer' [Aug 18 10:34:00] VERBOSE[12991] chan_sip.c: Scheduling destruction of SIP dialog '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' in 6400 ms (Method: INVITE) [Aug 18 10:34:00] DEBUG[12991] chan_sip.c: Strict routing enforced for session 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:00] VERBOSE[12991] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:00] DEBUG[12991] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:00] DEBUG[12991] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:00] VERBOSE[12991] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[12991] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[12991] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #51 [Aug 18 10:34:00] DEBUG[12991] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:00] DEBUG[13742] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13741] app.c: play_and_record: , /var/spool/asterisk/recording/213007_pPaTJtaVJwuRGbJSRCzOzmloUbPCTlFs, 'wav' [Aug 18 10:34:00] DEBUG[13741] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:00] VERBOSE[13741] app.c: x=0, open writing: /var/spool/asterisk/recording/213007_pPaTJtaVJwuRGbJSRCzOzmloUbPCTlFs format: wav, 0x7f0c70054fb0 [Aug 18 10:34:00] DEBUG[13720] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13742] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418;received=178.62.121.41 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:00] DEBUG[13720] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] VERBOSE[13235] app.c: User hung up [Aug 18 10:34:00] DEBUG[13235] res_stasis_recording.c: 1629282831.67: Recording complete [Aug 18 10:34:00] DEBUG[13235] channel.c: Channel 0x7f0c8c02ea90 'Recorder/ARI-00000005;1' hanging up. Refs: 2 [Aug 18 10:34:00] DEBUG[13742] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13736] http.c: match request [ari/channels/213057] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13738] http.c: match request [ari/channels/213059] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13743] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE add candidate: 10.131.0.10:19866, 2130706431 [Aug 18 10:34:00] DEBUG[13720] rtp_engine.c: RTP instance '0x7f0c2c08f640' is setup and ready to go [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE stopped [Aug 18 10:34:00] DEBUG[13720] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13720] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13720] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13720] res_rtp_asterisk.c: (0x7f0c2c08f640) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13720] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13728] http.c: match request [ari/channels/213055] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13234] channel.c: Channel 0x7f0c8c037210 'Recorder/ARI-00000005;2' destroying [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13738] http.c: match request [ari/channels/213059] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13738] http.c: match request [ari/channels/213059] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13743] http.c: HTTP Request URI is /ari/channels/213058?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116982&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13736] http.c: match request [ari/channels/213057] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13738] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[13711] channel.c: Channel 0x7f0c08044260 'Announcer/ARI-00000022;1' allocated [Aug 18 10:34:00] DEBUG[13711] stasis.c: Creating topic. name: channel:1629282840.192, detail: [Aug 18 10:34:00] DEBUG[13711] stasis.c: Topic 'channel:1629282840.192': 0x7f0c08086490 created [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP audio difference is 4824, ms is 623 [Aug 18 10:34:00] VERBOSE[13454] res_rtp_asterisk.c: 0x7f0c9800ef10 -- Strict RTP learning complete - Locking on source address 178.62.121.41:16938 [Aug 18 10:34:00] DEBUG[13720] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13711] stasis.c: Creating topic. name: cache:229/channel:1629282840.192, detail: [Aug 18 10:34:00] DEBUG[13711] stasis.c: Topic 'cache:229/channel:1629282840.192': 0x7f0c0804fc30 created [Aug 18 10:34:00] DEBUG[13742] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13234] stasis.c: Destroying topic. name: cache:80/channel:1629282831.68, detail: [Aug 18 10:34:00] DEBUG[13234] stasis.c: Topic 'cache:80/channel:1629282831.68': 0x7f0c8c01ff90 destroyed [Aug 18 10:34:00] DEBUG[13234] stasis.c: Destroying topic. name: channel:1629282831.68, detail: [Aug 18 10:34:00] DEBUG[13234] stasis.c: Topic 'channel:1629282831.68': 0x7f0c8c038f60 destroyed [Aug 18 10:34:00] DEBUG[13733] http.c: HTTP Request URI is /ari/playbacks/c40934e1-0986-4b52-804d-8eb899cc8791 [Aug 18 10:34:00] DEBUG[13728] http.c: match request [ari/channels/213055] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13715] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: Allocating new SIP dialog for 587979de31344d463b9ba7b463b7b18f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13723] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c28107140' [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) RTP allocated port 13928 [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE creating session 0.0.0.0:13928 (13928) [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE create [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE add system candidates [Aug 18 10:34:00] DEBUG[13723] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13723] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE add candidate: 159.65.48.104:13928, 2130706431 [Aug 18 10:34:00] DEBUG[13723] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13723] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13736] http.c: match request [ari/channels/213057] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116999@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625 Max-Forwards: 70 From: ;tag=as6ceaa437 To: Contact: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116999@178.62.121.41", nonce="048e446a", response="229b027e812dcf7d91509c351fa56509" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1137801089 1137801090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Session timer stopped: 11 - 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (3) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Session timer stopped: 14 - 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0;received=159.65.48.104 From: ;tag=as6093d024 To: ;tag=as49ef3a53 Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ce9240" Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5a56a1e0;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as49ef3a53 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ce9240" [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag as49ef3a53 [Aug 18 10:34:00] DEBUG[13742] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13742] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13742] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13742] stasis.c: Creating topic. name: bridge:beb17a84-adfc-4fa3-b7a8-31977a540c1f, detail: [Aug 18 10:34:00] DEBUG[13742] stasis.c: Topic 'bridge:beb17a84-adfc-4fa3-b7a8-31977a540c1f': 0x7f0c7c026f10 created [Aug 18 10:34:00] DEBUG[13742] stasis.c: Creating topic. name: cache:230/bridge:beb17a84-adfc-4fa3-b7a8-31977a540c1f, detail: [Aug 18 10:34:00] DEBUG[13742] stasis.c: Topic 'cache:230/bridge:beb17a84-adfc-4fa3-b7a8-31977a540c1f': 0x7f0c7c0110b0 created [Aug 18 10:34:00] DEBUG[13742] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13742] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13742] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13742] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:00] DEBUG[13742] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13742] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: calling simple_bridge technology constructor [Aug 18 10:34:00] DEBUG[13742] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: calling simple_bridge technology start [Aug 18 10:34:00] DEBUG[13738] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Finding handler for channels/213059 [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE add candidate: 10.131.0.10:13928, 2130706431 [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel Recorder/ARI-00000020;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:00] DEBUG[13702] channel.c: Channel Recorder/ARI-00000020;2 setting write format path: alaw -> slin [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13564] res_rtp_asterisk.c: (0x7f0c24077280) RTP audio difference is 864, ms is 74 [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: SIP call-id changed from '1413cf3f4a3226574abfed386569ce53@127.0.1.1:5060' to '78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13720] stasis.c: Creating topic. name: channel:213056, detail: [Aug 18 10:34:00] DEBUG[13720] stasis.c: Topic 'channel:213056': 0x7f0c2c07bc70 created [Aug 18 10:34:00] DEBUG[13720] stasis.c: Creating topic. name: cache:231/channel:213056, detail: [Aug 18 10:34:00] DEBUG[13720] stasis.c: Topic 'cache:231/channel:213056': 0x7f0c2c012da0 created [Aug 18 10:34:00] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13723] rtp_engine.c: RTP instance '0x7f0c28107140' is setup and ready to go [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) ICE stopped [Aug 18 10:34:00] DEBUG[13723] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13723] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13723] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13723] res_rtp_asterisk.c: (0x7f0c28107140) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13723] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13723] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: SIP call-id changed from '587979de31344d463b9ba7b463b7b18f@127.0.1.1:5060' to '2596122845f5f4322466678f68967bbf@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13723] stasis.c: Creating topic. name: channel:213054, detail: [Aug 18 10:34:00] DEBUG[13723] stasis.c: Topic 'channel:213054': 0x7f0c280d5d40 created [Aug 18 10:34:00] DEBUG[13723] stasis.c: Creating topic. name: cache:232/channel:213054, detail: [Aug 18 10:34:00] DEBUG[13723] stasis.c: Topic 'cache:232/channel:213054': 0x7f0c280d6770 created [Aug 18 10:34:00] VERBOSE[13739] dial.c: Called 127.0.0.1:50118 [Aug 18 10:34:00] VERBOSE[13739] dial.c: UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 answered [Aug 18 10:34:00] VERBOSE[13739] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 [Aug 18 10:34:00] DEBUG[13739] stasis/app.c: Channel 'robot_213008' is 2 interested in calls_0 [Aug 18 10:34:00] DEBUG[13736] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13728] http.c: match request [ari/channels/213055] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13250] bridge_channel.c: Setting 0x7f0c9c021fe0(Snoop/212986-00000002) state from:0 to:1 [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13728] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13742] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:00] DEBUG[13736] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Finding handler for 213059 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Stopping retransmission on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:00] DEBUG[13742] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking channels create: Didn't match 213059 [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Finding handler for channels/213057 [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Finding handler for 213057 [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking channels create: Didn't match 213057 [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13736] res_ari.c: Checking channels externalMedia: Didn't match 213057 [Aug 18 10:34:00] DEBUG[13736] res_ari.c: No explicit handler found for 213057. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13738] res_ari.c: Checking channels externalMedia: Didn't match 213059 [Aug 18 10:34:00] DEBUG[13738] res_ari.c: No explicit handler found for 213059. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13250] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: pulling 0x7f0c9c021fe0(Snoop/212986-00000002) [Aug 18 10:34:00] VERBOSE[13250] bridge_channel.c: Channel Snoop/212986-00000002 left 'simple_bridge' stasis-bridge <8b092052-108a-4921-8aad-1aecb4e2c824> [Aug 18 10:34:00] DEBUG[13250] bridge_channel.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: 0x7f0c9c021fe0(Snoop/212986-00000002) is leaving simple_bridge technology [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13686] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] VERBOSE[13459] res_rtp_asterisk.c: 0x7f0c94023d30 -- Strict RTP learning complete - Locking on source address 178.62.121.41:13636 [Aug 18 10:34:00] DEBUG[13728] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Finding handler for channels/213055 [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Finding handler for 213055 [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking channels create: Didn't match 213055 [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13728] res_ari.c: Checking channels externalMedia: Didn't match 213055 [Aug 18 10:34:00] DEBUG[13728] res_ari.c: No explicit handler found for 213055. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 1296, ms is 101 [Aug 18 10:34:00] DEBUG[13743] http.c: match request [ari/channels/213058] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Setting 0x7f0c7c01ea60(SIP/zvonobot-00000009) state from:0 to:1 [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13744] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13743] http.c: match request [ari/channels/213058] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13747] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 656, ms is 61 [Aug 18 10:34:00] DEBUG[13250] bridge_native_rtp.c: Bridge '8b092052-108a-4921-8aad-1aecb4e2c824' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13746] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13250] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13733] http.c: match request [ari/playbacks/c40934e1-0986-4b52-804d-8eb899cc8791] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13250] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13250] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13733] http.c: match request [ari/playbacks/c40934e1-0986-4b52-804d-8eb899cc8791] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13746] http.c: HTTP Request URI is /ari/channels/213007/snoop?app=calls_0&spy=in [Aug 18 10:34:00] DEBUG[13744] http.c: HTTP Request URI is /ari/channels/213060?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116980&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13250] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13250] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824 is already using the new technology. [Aug 18 10:34:00] DEBUG[13715] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (3) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:00] DEBUG[13747] http.c: HTTP Request URI is /ari/channels/213061?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116979&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13704] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13704] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13704] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13704] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13704] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13704] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677 is already using the new technology. [Aug 18 10:34:00] DEBUG[13704] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[13748] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13743] http.c: match request [ari/channels/213058] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:00] DEBUG[13743] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 792, ms is 119 [Aug 18 10:34:00] DEBUG[13750] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13750] http.c: HTTP Request URI is /ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/record?name=212982_SjNdOhkmKEHHUVlGZbmvYPyClcnqXyTA&format=wav [Aug 18 10:34:00] DEBUG[13750] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/record] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13750] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/record] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13750] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/record] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13750] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Finding handler for bridges/45640e14-e267-477d-81ea-fbac374f9677/record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Finding handler for 45640e14-e267-477d-81ea-fbac374f9677 [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13750] res_ari.c: No explicit handler found for 45640e14-e267-477d-81ea-fbac374f9677. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Finding handler for record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:00] DEBUG[13750] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:00] DEBUG[13746] http.c: match request [ari/channels/213007/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13697] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13697] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13744] http.c: match request [ari/channels/213060] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13748] http.c: HTTP Request URI is /ari/channels/213063?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116977&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13746] http.c: match request [ari/channels/213007/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pulling 0x7f0c7c01ea60(SIP/zvonobot-00000009) [Aug 18 10:34:00] DEBUG[13746] http.c: match request [ari/channels/213007/snoop] with handler [ari] len 3 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] VERBOSE[13349] bridge_channel.c: Channel SIP/zvonobot-00000009 left 'softmix' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:34:00] DEBUG[13751] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13564] res_rtp_asterisk.c: (0x7f0c24077280) RTP audio difference is 832, ms is 72 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13751] http.c: HTTP Request URI is /ari/channels/213062?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116978&callerId=74950493843 [Aug 18 10:34:00] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP changing ssrc from 1138480681 to 531643435 due to a source change [Aug 18 10:34:00] DEBUG[13744] http.c: match request [ari/channels/213060] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13748] http.c: match request [ari/channels/213063] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13714] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c7c01ea60(SIP/zvonobot-00000009) is leaving softmix technology [Aug 18 10:34:00] DEBUG[12971] stasis/app.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' is 2 interested in calls_0 [Aug 18 10:34:00] DEBUG[13746] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13715] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Setting 0x7f0c200534f0(Announcer/ARI-00000016;2) state from:0 to:2 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (3) INVITE - 5 [Aug 18 10:34:00] DEBUG[13748] http.c: match request [ari/channels/213063] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13714] netsock2.c: Splitting '127.0.0.1:50194' into... [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Finding handler for channels/213007/snoop [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:00] DEBUG[13714] netsock2.c: ...host '127.0.0.1' and port '50194'. [Aug 18 10:34:00] DEBUG[13748] http.c: match request [ari/channels/213063] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13733] http.c: match request [ari/playbacks/c40934e1-0986-4b52-804d-8eb899cc8791] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13747] http.c: match request [ari/channels/213061] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13743] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13744] http.c: match request [ari/channels/213060] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13748] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13706] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13714] netsock2.c: Splitting '127.0.0.1:50194' into... [Aug 18 10:34:00] DEBUG[13714] netsock2.c: ...host '127.0.0.1' and port '50194'. [Aug 18 10:34:00] DEBUG[13714] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:00] DEBUG[13714] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c0b2b20' [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP allocated port 16388 [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE creating session 127.0.0.1:16388 (16388) [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE create [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE add system candidates [Aug 18 10:34:00] DEBUG[13714] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13714] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE add candidate: 159.65.48.104:16388, 2130706431 [Aug 18 10:34:00] DEBUG[13714] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13714] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13714] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE add candidate: 10.131.0.10:16388, 2130706431 [Aug 18 10:34:00] DEBUG[13714] rtp_engine.c: RTP instance '0x7f0c1c0b2b20' is setup and ready to go [Aug 18 10:34:00] DEBUG[13714] stasis.c: Creating topic. name: channel:robot_212977, detail: [Aug 18 10:34:00] DEBUG[13714] stasis.c: Topic 'channel:robot_212977': 0x7f0c1c043530 created [Aug 18 10:34:00] DEBUG[13714] stasis.c: Creating topic. name: cache:233/channel:robot_212977, detail: [Aug 18 10:34:00] DEBUG[13714] stasis.c: Topic 'cache:233/channel:robot_212977': 0x7f0c1c043c80 created [Aug 18 10:34:00] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13747] http.c: match request [ari/channels/213061] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13748] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13733] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: Allocating new SIP dialog for 7f7cbdff141d1e8c6d35960873a1034d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13738] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c40073870' [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) RTP allocated port 11858 [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE creating session 0.0.0.0:11858 (11858) [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE create [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE add system candidates [Aug 18 10:34:00] DEBUG[13738] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13738] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE add candidate: 159.65.48.104:11858, 2130706431 [Aug 18 10:34:00] DEBUG[13738] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13738] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE add candidate: 10.131.0.10:11858, 2130706431 [Aug 18 10:34:00] DEBUG[13738] rtp_engine.c: RTP instance '0x7f0c40073870' is setup and ready to go [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) ICE stopped [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Finding handler for 213007 [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channels create: Didn't match 213007 [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channels externalMedia: Didn't match 213007 [Aug 18 10:34:00] DEBUG[13746] res_ari.c: No explicit handler found for 213007. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Finding handler for snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:00] DEBUG[13746] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:00] DEBUG[13750] stasis.c: Creating topic. name: channel:1629282840.195, detail: [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Finding handler for channels/213063 [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Finding handler for 213063 [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking channels create: Didn't match 213063 [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13748] res_ari.c: Checking channels externalMedia: Didn't match 213063 [Aug 18 10:34:00] DEBUG[13748] res_ari.c: No explicit handler found for 213063. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13747] http.c: match request [ari/channels/213061] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13250] bridge_channel.c: Bridge is returning 0x7f0c9c021fe0(Snoop/212986-00000002) to read format slin [Aug 18 10:34:00] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13751] http.c: match request [ari/channels/213062] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13751] http.c: match request [ari/channels/213062] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: Allocating new SIP dialog for 151c5e806a1028a9549b34e40573e414@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13736] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38087180' [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) RTP allocated port 13840 [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE creating session 0.0.0.0:13840 (13840) [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE create [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE add system candidates [Aug 18 10:34:00] DEBUG[13736] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13736] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13738] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13738] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13738] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13738] res_rtp_asterisk.c: (0x7f0c40073870) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13738] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13738] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13738] chan_sip.c: SIP call-id changed from '7f7cbdff141d1e8c6d35960873a1034d@127.0.1.1:5060' to '18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13738] stasis.c: Creating topic. name: channel:213059, detail: [Aug 18 10:34:00] DEBUG[13738] stasis.c: Topic 'channel:213059': 0x7f0c40070e10 created [Aug 18 10:34:00] DEBUG[13738] stasis.c: Creating topic. name: cache:234/channel:213059, detail: [Aug 18 10:34:00] DEBUG[13738] stasis.c: Topic 'cache:234/channel:213059': 0x7f0c40071840 created [Aug 18 10:34:00] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:00] DEBUG[13250] channel.c: Channel Snoop/212986-00000002 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13250] bridge_channel.c: Bridge is returning 0x7f0c9c021fe0(Snoop/212986-00000002) to write format slin [Aug 18 10:34:00] DEBUG[13747] http.c: Match made with [ari] [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13751] http.c: match request [ari/channels/213062] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13250] channel.c: Channel Snoop/212986-00000002 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE add candidate: 159.65.48.104:13840, 2130706431 [Aug 18 10:34:00] DEBUG[13736] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13736] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE add candidate: 10.131.0.10:13840, 2130706431 [Aug 18 10:34:00] DEBUG[13736] rtp_engine.c: RTP instance '0x7f0c38087180' is setup and ready to go [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) ICE stopped [Aug 18 10:34:00] DEBUG[13736] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13736] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13736] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13736] res_rtp_asterisk.c: (0x7f0c38087180) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13736] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13736] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13736] chan_sip.c: SIP call-id changed from '151c5e806a1028a9549b34e40573e414@127.0.1.1:5060' to '1ee655842d2ed684574010b3091c860a@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13736] stasis.c: Creating topic. name: channel:213057, detail: [Aug 18 10:34:00] DEBUG[13736] stasis.c: Topic 'channel:213057': 0x7f0c38058250 created [Aug 18 10:34:00] DEBUG[13736] stasis.c: Creating topic. name: cache:235/channel:213057, detail: [Aug 18 10:34:00] DEBUG[13736] stasis.c: Topic 'cache:235/channel:213057': 0x7f0c38058c80 created [Aug 18 10:34:00] DEBUG[13747] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13250] stasis/control.c: 1629282831.69, 8b092052-108a-4921-8aad-1aecb4e2c824: Channel was departed from bridge [Aug 18 10:34:00] DEBUG[13751] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000009 - start 1629282822.267188 answer 1629282833.290648 end 1629282840.313133 dur 18.045 bill 7.022 dispo ANSWERED [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #64 (3) INVITE - 5 [Aug 18 10:34:00] DEBUG[13250] stasis/app.c: bridge '8b092052-108a-4921-8aad-1aecb4e2c824': is 2 interested in calls_0 [Aug 18 10:34:00] DEBUG[13349] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e'. Checking compatability for channels 'Announcer/ARI-00000016;2' and 'Recorder/ARI-0000000d;2' [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Finding handler for channels/213058 [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13250] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:00] DEBUG[13349] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as could not get details [Aug 18 10:34:00] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 1104, ms is 89 [Aug 18 10:34:00] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 1088, ms is 88 [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13349] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] VERBOSE[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: switching from softmix technology to simple_bridge [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology constructor [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c200534f0(Announcer/ARI-00000016;2) to dummy bridge temporarily [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: moving 0x7f0c240520b0(Recorder/ARI-0000000d;2) to dummy bridge temporarily [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c200534f0(Announcer/ARI-00000016;2) is leaving softmix technology (dummy) [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is leaving softmix technology (dummy) [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology stop [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c200534f0(Announcer/ARI-00000016;2) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Recorder/ARI-0000000d;2 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Announcer/ARI-00000016;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Finding handler for playbacks/c40934e1-0986-4b52-804d-8eb899cc8791 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13238] stasis/control.c: 1629282831.69: Channel departing bridge [Aug 18 10:34:00] DEBUG[13238] bridge.c: Waiting for 0x7f0c9c021fe0(Snoop/212986-00000002) bridge thread to die. [Aug 18 10:34:00] DEBUG[13238] stasis/app.c: channel '1629282831.69': is 0 interested in calls_0 [Aug 18 10:34:00] DEBUG[13238] stasis/app.c: channel '1629282831.69' unsubscribed from calls_0 [Aug 18 10:34:00] DEBUG[13238] channel.c: Channel 0x7f0ca8007d50 'Snoop/212986-00000002' hanging up. Refs: 3 [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Announcer/ARI-00000016;2 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 864, ms is 74 [Aug 18 10:34:00] DEBUG[13750] stasis.c: Topic 'channel:1629282840.195': 0x7f0c94064810 created [Aug 18 10:34:00] DEBUG[13751] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #64)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Finding handler for playbacks [Aug 18 10:34:00] DEBUG[13716] stasis.c: Creating topic. name: channel:1629282840.199, detail: [Aug 18 10:34:00] DEBUG[13716] stasis.c: Topic 'channel:1629282840.199': 0x7f0c18091270 created [Aug 18 10:34:00] DEBUG[13716] stasis.c: Creating topic. name: cache:237/channel:1629282840.199, detail: [Aug 18 10:34:00] DEBUG[13716] stasis.c: Topic 'cache:237/channel:1629282840.199': 0x7f0c180b95f0 created [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Finding handler for 213058 [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking channels create: Didn't match 213058 [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13743] res_ari.c: Checking channels externalMedia: Didn't match 213058 [Aug 18 10:34:00] DEBUG[13743] res_ari.c: No explicit handler found for 213058. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13744] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13750] stasis.c: Creating topic. name: cache:236/channel:1629282840.195, detail: [Aug 18 10:34:00] DEBUG[13750] stasis.c: Topic 'cache:236/channel:1629282840.195': 0x7f0c94030e90 created [Aug 18 10:34:00] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: Allocating new SIP dialog for 3eaac61608f992311ad6454d6dde468a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13728] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c340ed1d0' [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) RTP allocated port 14460 [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE creating session 0.0.0.0:14460 (14460) [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE create [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE add system candidates [Aug 18 10:34:00] DEBUG[13728] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13728] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE add candidate: 159.65.48.104:14460, 2130706431 [Aug 18 10:34:00] DEBUG[13728] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13728] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE add candidate: 10.131.0.10:14460, 2130706431 [Aug 18 10:34:00] DEBUG[13728] rtp_engine.c: RTP instance '0x7f0c340ed1d0' is setup and ready to go [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE stopped [Aug 18 10:34:00] DEBUG[13728] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13728] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13728] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13728] res_rtp_asterisk.c: (0x7f0c340ed1d0) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13728] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13728] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13728] chan_sip.c: SIP call-id changed from '3eaac61608f992311ad6454d6dde468a@127.0.1.1:5060' to '00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13728] stasis.c: Creating topic. name: channel:213055, detail: [Aug 18 10:34:00] DEBUG[13728] stasis.c: Topic 'channel:213055': 0x7f0c340fdce0 created [Aug 18 10:34:00] DEBUG[13728] stasis.c: Creating topic. name: cache:238/channel:213055, detail: [Aug 18 10:34:00] DEBUG[13728] stasis.c: Topic 'cache:238/channel:213055': 0x7f0c340fe760 created [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Finding handler for channels/213062 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Finding handler for channels/213061 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Recorder/ARI-0000000d;2 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Announcer/ARI-00000016;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Announcer/ARI-00000016;2 setting read format path: slin -> slin [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel Recorder/ARI-0000000d;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology start [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: deferring softmix technology destructor [Aug 18 10:34:00] DEBUG[13349] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: queueing action type:13 sub:1000 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:00] DEBUG[13701] channel.c: Soft-Hanging (0x20) up channel 'Snoop/212986-00000002' [Aug 18 10:34:00] DEBUG[13701] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Finding handler for 213062 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Finding handler for c40934e1-0986-4b52-804d-8eb899cc8791 [Aug 18 10:34:00] DEBUG[13733] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13733] res_ari.c: No explicit handler found for c40934e1-0986-4b52-804d-8eb899cc8791. Using wildcard playbackId. [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking channels create: Didn't match 213062 [Aug 18 10:34:00] DEBUG[13701] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13733] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13733] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 992, ms is 82 [Aug 18 10:34:00] DEBUG[13732] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:00] DEBUG[13732] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:00] DEBUG[13732] channel.c: Channel Announcer/ARI-0000001f;1 setting write format path: slin -> slin [Aug 18 10:34:00] NOTICE[13732] res_stasis_playback.c: 1629282839.179: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:00] DEBUG[13732] channel.c: Channel 0x7f0c240f5d70 'Announcer/ARI-0000001f;1' hanging up. Refs: 2 [Aug 18 10:34:00] DEBUG[13744] http.c: HTTP consuming request body [Aug 18 10:34:00] DEBUG[13751] res_ari.c: Checking channels externalMedia: Didn't match 213062 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116997@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081 Max-Forwards: 70 From: ;tag=as0e0b214d To: Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116997@178.62.121.41", nonce="2dbc5b08", response="c92db07f4aae82ff2863950559637b69" Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1275485183 1275485184 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[13753] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13751] res_ari.c: No explicit handler found for 213062. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13717] channel.c: Channel 0x7f0c2008a2f0 'Recorder/ARI-00000023;1' allocated [Aug 18 10:34:00] DEBUG[13717] stasis.c: Creating topic. name: channel:1629282840.201, detail: [Aug 18 10:34:00] DEBUG[13717] stasis.c: Topic 'channel:1629282840.201': 0x7f0c2006f040 created [Aug 18 10:34:00] DEBUG[13717] stasis.c: Creating topic. name: cache:239/channel:1629282840.201, detail: [Aug 18 10:34:00] DEBUG[13717] stasis.c: Topic 'cache:239/channel:1629282840.201': 0x7f0c20089a60 created [Aug 18 10:34:00] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13752] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13706] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13753] http.c: HTTP Request URI is /ari/channels/robot_212999 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Finding handler for channels/213060 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP got report of 100 bytes from 178.62.121.41:10913 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13753] http.c: match request [ari/channels/robot_212999] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13752] http.c: HTTP Request URI is /ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:34:00] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13753] http.c: match request [ari/channels/robot_212999] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[13564] res_rtp_asterisk.c: (0x7f0c24077280) RTP audio difference is 720, ms is 65 [Aug 18 10:34:00] DEBUG[13752] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13507] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: stopping mixing thread [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:00] DEBUG[20534] bridge_softmix.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: Waiting for mixing thread to die. [Aug 18 10:34:00] DEBUG[13503] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pulling 0x7f0c200534f0(Announcer/ARI-00000016;2) [Aug 18 10:34:00] DEBUG[13753] http.c: match request [ari/channels/robot_212999] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13706] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[13509] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] VERBOSE[13503] bridge_channel.c: Channel Announcer/ARI-00000016;2 left 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:34:00] DEBUG[13503] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c200534f0(Announcer/ARI-00000016;2) is leaving simple_bridge technology [Aug 18 10:34:00] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP creating BEGIN DTMF Frame: 51 (3), at 178.62.121.41:10224 [Aug 18 10:34:00] DEBUG[13503] bridge_channel.c: Setting 0x7f0c240520b0(Recorder/ARI-0000000d;2) state from:0 to:2 [Aug 18 10:34:00] DTMF[13627] channel.c: DTMF begin '3' received on SIP/zvonobot-0000003b [Aug 18 10:34:00] DTMF[13627] channel.c: DTMF begin passthrough '3' on SIP/zvonobot-0000003b [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Finding handler for 213060 [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: Allocating new SIP dialog for 0c8839d03fa5bd2e56c3b5d91fbd6f98@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13743] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7806cff0' [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) RTP allocated port 11586 [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE creating session 0.0.0.0:11586 (11586) [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE create [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add system candidates [Aug 18 10:34:00] DEBUG[13743] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13743] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add candidate: 159.65.48.104:11586, 2130706431 [Aug 18 10:34:00] DEBUG[13743] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13743] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add candidate: 10.131.0.10:11586, 2130706431 [Aug 18 10:34:00] DEBUG[13743] rtp_engine.c: RTP instance '0x7f0c7806cff0' is setup and ready to go [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE stopped [Aug 18 10:34:00] DEBUG[13743] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13743] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13743] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13743] res_rtp_asterisk.c: (0x7f0c7806cff0) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13743] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13743] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13753] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13752] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13752] http.c: match request [ari/bridges/90245cdf-0ee9-4414-b99e-c22349f119a2] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13628] res_stasis_recording.c: 1629282838.167: Sending record(213011_NLuXdhmeVPfdDxEObkwOLqvNJGIULStr.wav) command [Aug 18 10:34:00] DEBUG[13628] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:00] DEBUG[13628] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 445 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 445 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:00] DEBUG[13757] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking channels create: Didn't match 213060 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Finding handler for 213061 [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking channels create: Didn't match 213061 [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13747] res_ari.c: Checking channels externalMedia: Didn't match 213061 [Aug 18 10:34:00] DEBUG[13747] res_ari.c: No explicit handler found for 213061. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13754] app.c: play_and_record: , /var/spool/asterisk/recording/213011_NLuXdhmeVPfdDxEObkwOLqvNJGIULStr, 'wav' [Aug 18 10:34:00] DEBUG[13754] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:00] VERBOSE[13754] app.c: x=0, open writing: /var/spool/asterisk/recording/213011_NLuXdhmeVPfdDxEObkwOLqvNJGIULStr format: wav, 0x7f0c980a7c50 [Aug 18 10:34:00] DEBUG[13757] http.c: HTTP Request URI is /ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a/addChannel?channel=1629282839.182%2Crobot_213008 [Aug 18 10:34:00] DEBUG[13744] res_ari.c: Checking channels externalMedia: Didn't match 213060 [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DTMF[13698] channel.c: DTMF begin '3' received on Recorder/ARI-0000001e;1 [Aug 18 10:34:00] DTMF[13698] channel.c: DTMF begin passthrough '3' on Recorder/ARI-0000001e;1 [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Bridge is returning 0x7f0c7c01ea60(SIP/zvonobot-00000009) to read format alaw [Aug 18 10:34:00] DEBUG[13744] res_ari.c: No explicit handler found for 213060. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[13685] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 From: ;tag=as30c9fc1b To: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13752] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel SIP/zvonobot-00000009 setting read format path: alaw -> alaw [Aug 18 10:34:00] DEBUG[13756] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13756] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:00] DEBUG[13503] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13757] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13349] bridge_channel.c: Bridge is returning 0x7f0c7c01ea60(SIP/zvonobot-00000009) to write format alaw [Aug 18 10:34:00] DEBUG[13685] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Finding handler for bridges/90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:34:00] DEBUG[13503] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13349] channel.c: Channel SIP/zvonobot-00000009 setting write format path: alaw -> alaw [Aug 18 10:34:00] DEBUG[13349] stasis/control.c: 212973, 85c47f7b-0e24-4408-b7bf-5a532802bd8e: Channel was departed from bridge [Aug 18 10:34:00] DEBUG[13349] stasis/app.c: bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e': is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13349] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:00] DEBUG[12903] stasis/control.c: 212973: Channel departing bridge [Aug 18 10:34:00] DEBUG[12903] bridge.c: Waiting for 0x7f0c7c01ea60(SIP/zvonobot-00000009) bridge thread to die. [Aug 18 10:34:00] DEBUG[12903] stasis/app.c: channel '212973': is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK33cd0033;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[13757] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[12903] channel.c: Channel 0x7f0c700195c0 'SIP/zvonobot-00000009' hanging up. Refs: 3 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Finding handler for channels/robot_212999 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as30c9fc1b [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060 (Checking To) --From tag as30c9fc1b --To-tag [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Finding handler for 90245cdf-0ee9-4414-b99e-c22349f119a2 [Aug 18 10:34:00] DEBUG[13752] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13752] res_ari.c: No explicit handler found for 90245cdf-0ee9-4414-b99e-c22349f119a2. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13752] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: telling all channels to leave the party [Aug 18 10:34:00] DEBUG[13752] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:00] DEBUG[13752] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: queueing action type:13 sub:1001 [Aug 18 10:34:00] DEBUG[13752] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13743] chan_sip.c: SIP call-id changed from '0c8839d03fa5bd2e56c3b5d91fbd6f98@127.0.1.1:5060' to '2cc23538293c1849651dca44558c8447@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13743] stasis.c: Creating topic. name: channel:213058, detail: [Aug 18 10:34:00] DEBUG[13743] stasis.c: Topic 'channel:213058': 0x7f0c7803c810 created [Aug 18 10:34:00] DEBUG[13743] stasis.c: Creating topic. name: cache:240/channel:213058, detail: [Aug 18 10:34:00] DEBUG[13743] stasis.c: Topic 'cache:240/channel:213058': 0x7f0c7803abb0 created [Aug 18 10:34:00] DEBUG[13752] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13503] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13503] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13503] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13503] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e is already using the new technology. [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '68132887044a819b71fb501f4d1e2ab3@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:00] DEBUG[13706] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13759] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13759] http.c: HTTP Request URI is /ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (3) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: Allocating new SIP dialog for 7f953ed1633f6daa4c05d2ee361a4aa8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13747] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c072bb0' [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) RTP allocated port 14668 [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE creating session 0.0.0.0:14668 (14668) [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE create [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE add system candidates [Aug 18 10:34:00] DEBUG[13747] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13747] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE add candidate: 159.65.48.104:14668, 2130706431 [Aug 18 10:34:00] DEBUG[13747] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13747] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE add candidate: 10.131.0.10:14668, 2130706431 [Aug 18 10:34:00] DEBUG[13747] rtp_engine.c: RTP instance '0x7f0c8c072bb0' is setup and ready to go [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE stopped [Aug 18 10:34:00] DEBUG[13747] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13747] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13747] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13747] res_rtp_asterisk.c: (0x7f0c8c072bb0) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13747] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13757] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a/addChannel] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: Allocating new SIP dialog for 0d56aaf322ddc7e054054ee44e352d1b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13751] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c90052d80' [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) RTP allocated port 17282 [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE creating session 0.0.0.0:17282 (17282) [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE create [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE add system candidates [Aug 18 10:34:00] DEBUG[13751] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13751] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE add candidate: 159.65.48.104:17282, 2130706431 [Aug 18 10:34:00] DEBUG[13751] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13751] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE add candidate: 10.131.0.10:17282, 2130706431 [Aug 18 10:34:00] DEBUG[13751] rtp_engine.c: RTP instance '0x7f0c90052d80' is setup and ready to go [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) ICE stopped [Aug 18 10:34:00] DEBUG[13751] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13751] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13751] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13751] res_rtp_asterisk.c: (0x7f0c90052d80) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13751] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13751] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13751] chan_sip.c: SIP call-id changed from '0d56aaf322ddc7e054054ee44e352d1b@127.0.1.1:5060' to '083441fd621bd040753e952c5d9a1860@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13751] stasis.c: Creating topic. name: channel:213062, detail: [Aug 18 10:34:00] DEBUG[13751] stasis.c: Topic 'channel:213062': 0x7f0c90080810 created [Aug 18 10:34:00] DEBUG[13751] stasis.c: Creating topic. name: cache:241/channel:213062, detail: [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #51 (1) BYE - 8 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #51)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: Allocating new SIP dialog for 61905eac49658a983b96d9ec4f27a115@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13757] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[13758] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:00] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13751] stasis.c: Topic 'cache:241/channel:213062': 0x7f0c90081290 created [Aug 18 10:34:00] DEBUG[13353] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: pulling 0x7f0c240520b0(Recorder/ARI-0000000d;2) [Aug 18 10:34:00] DEBUG[13235] channel.c: Channel 0x7f0c8c02ea90 'Recorder/ARI-00000005;1' destroying [Aug 18 10:34:00] DEBUG[13756] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:00] VERBOSE[13353] bridge_channel.c: Channel Recorder/ARI-0000000d;2 left 'simple_bridge' stasis-bridge <85c47f7b-0e24-4408-b7bf-5a532802bd8e> [Aug 18 10:34:00] DEBUG[13353] bridge_channel.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: 0x7f0c240520b0(Recorder/ARI-0000000d;2) is leaving simple_bridge technology [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13758] http.c: HTTP Request URI is /ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13748] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c88063530' [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) RTP allocated port 16058 [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE creating session 0.0.0.0:16058 (16058) [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE create [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE add system candidates [Aug 18 10:34:00] DEBUG[13748] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13748] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE add candidate: 159.65.48.104:16058, 2130706431 [Aug 18 10:34:00] DEBUG[13748] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13748] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE add candidate: 10.131.0.10:16058, 2130706431 [Aug 18 10:34:00] DEBUG[13748] rtp_engine.c: RTP instance '0x7f0c88063530' is setup and ready to go [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) ICE stopped [Aug 18 10:34:00] DEBUG[13748] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13748] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13748] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13748] res_rtp_asterisk.c: (0x7f0c88063530) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13748] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13748] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13748] chan_sip.c: SIP call-id changed from '61905eac49658a983b96d9ec4f27a115@127.0.1.1:5060' to '7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13748] stasis.c: Creating topic. name: channel:213063, detail: [Aug 18 10:34:00] DEBUG[13748] stasis.c: Topic 'channel:213063': 0x7f0c8809dca0 created [Aug 18 10:34:00] DEBUG[13748] stasis.c: Creating topic. name: cache:242/channel:213063, detail: [Aug 18 10:34:00] DEBUG[13748] stasis.c: Topic 'cache:242/channel:213063': 0x7f0c8809e6d0 created [Aug 18 10:34:00] DEBUG[13686] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:00] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP creating END DTMF Frame: 51 (3), at 178.62.121.41:10224 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:34:00] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:34:00] DEBUG[13353] bridge_native_rtp.c: Bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13353] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13353] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13353] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13353] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13353] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e is already using the new technology. [Aug 18 10:34:00] DEBUG[13691] channel.c: Channel 0x7f0c90044400 'Announcer/ARI-00000021;2' allocated [Aug 18 10:34:00] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '90245cdf-0ee9-4414-b99e-c22349f119a2': is 0 interested in calls_0 [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '90245cdf-0ee9-4414-b99e-c22349f119a2' unsubscribed from calls_0 [Aug 18 10:34:00] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP got report of 100 bytes from 178.62.121.41:10789 [Aug 18 10:34:00] DEBUG[13503] channel.c: Channel 0x7f0c20061cb0 'Announcer/ARI-00000016;2' hanging up. Refs: 2 [Aug 18 10:34:00] DEBUG[13759] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824] with handler [httpstatus] len 10 [Aug 18 10:34:00] DTMF[13627] channel.c: DTMF end '3' received on SIP/zvonobot-0000003b, duration 100 ms [Aug 18 10:34:00] DEBUG[13758] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59] with handler [httpstatus] len 10 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 From: ;tag=as2b432b30 To: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling stasis bridge destructor [Aug 18 10:34:00] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:00] DEBUG[13678] channel.c: Channel Recorder/ARI-0000001c;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:00] DEBUG[13678] channel.c: Channel Recorder/ARI-0000001c;2 setting write format path: alaw -> slin [Aug 18 10:34:00] DEBUG[13454] audiohook.c: Audiohook 0x7f0ca005b420 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[13756] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13235] stasis.c: Destroying topic. name: cache:79/channel:1629282831.67, detail: [Aug 18 10:34:00] DEBUG[13235] stasis.c: Topic 'cache:79/channel:1629282831.67': 0x7f0c8c030cf0 destroyed [Aug 18 10:34:00] DEBUG[13235] stasis.c: Destroying topic. name: channel:1629282831.67, detail: [Aug 18 10:34:00] DEBUG[13235] stasis.c: Topic 'channel:1629282831.67': 0x7f0c8c018b60 destroyed [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 494 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 494 [Aug 18 10:34:00] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 656, ms is 61 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:00] DTMF[13627] channel.c: DTMF end accepted with begin '3' on SIP/zvonobot-0000003b [Aug 18 10:34:00] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Snoop - 212993 [Aug 18 10:34:00] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 1024, ms is 84 [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: Allocating new SIP dialog for 4a45c39d41b114ed0af802e2313edd91@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:00] DEBUG[13744] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c84090800' [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) RTP allocated port 12964 [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE creating session 0.0.0.0:12964 (12964) [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE create [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE add system candidates [Aug 18 10:34:00] DEBUG[13744] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:00] DEBUG[13744] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE add candidate: 159.65.48.104:12964, 2130706431 [Aug 18 10:34:00] DEBUG[13744] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:00] DEBUG[13744] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE add candidate: 10.131.0.10:12964, 2130706431 [Aug 18 10:34:00] DEBUG[13744] rtp_engine.c: RTP instance '0x7f0c84090800' is setup and ready to go [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) ICE stopped [Aug 18 10:34:00] DEBUG[13744] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:00] DEBUG[13744] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:00] DEBUG[13744] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:00] DEBUG[13744] res_rtp_asterisk.c: (0x7f0c84090800) RTCP setup on RTP instance [Aug 18 10:34:00] VERBOSE[13744] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13744] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13744] chan_sip.c: SIP call-id changed from '4a45c39d41b114ed0af802e2313edd91@127.0.1.1:5060' to '195b29ec6362148262de07066ce29e57@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13744] stasis.c: Creating topic. name: channel:213060, detail: [Aug 18 10:34:00] DEBUG[13744] stasis.c: Topic 'channel:213060': 0x7f0c841050e0 created [Aug 18 10:34:00] DEBUG[13744] stasis.c: Creating topic. name: cache:243/channel:213060, detail: [Aug 18 10:34:00] DEBUG[13744] stasis.c: Topic 'cache:243/channel:213060': 0x7f0c84105b60 created [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Finding handler for robot_212999 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking channels create: Didn't match robot_212999 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13753] res_ari.c: Checking channels externalMedia: Didn't match robot_212999 [Aug 18 10:34:00] DEBUG[13753] res_ari.c: No explicit handler found for robot_212999. Using wildcard channelId. [Aug 18 10:34:00] DEBUG[20535] devicestate.c: Changing state for Snoop/212993 - state 4 (Invalid) [Aug 18 10:34:00] DEBUG[13756] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DTMF[13627] channel.c: DTMF end passthrough '3' on SIP/zvonobot-0000003b [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[13691] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:00] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 90245cdf-0ee9-4414-b99e-c22349f119a2: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:00] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:00] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13759] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13759] http.c: match request [ari/bridges/8b092052-108a-4921-8aad-1aecb4e2c824] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: cache:77/bridge:90245cdf-0ee9-4414-b99e-c22349f119a2, detail: [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'cache:77/bridge:90245cdf-0ee9-4414-b99e-c22349f119a2': 0x7f0c7c017490 destroyed [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: bridge:90245cdf-0ee9-4414-b99e-c22349f119a2, detail: [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'bridge:90245cdf-0ee9-4414-b99e-c22349f119a2': 0x7f0c7c0068a0 destroyed [Aug 18 10:34:00] DEBUG[13691] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000021;1' [Aug 18 10:34:00] DEBUG[13760] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c90058340(Announcer/ARI-00000021;2) is joining [Aug 18 10:34:00] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/Snoop/212993, detail: [Aug 18 10:34:00] DEBUG[13353] channel.c: Channel 0x7f0c24056340 'Recorder/ARI-0000000d;2' hanging up. Refs: 2 [Aug 18 10:34:00] DTMF[13698] channel.c: DTMF end '3' received on Recorder/ARI-0000001e;1, duration 100 ms [Aug 18 10:34:00] DTMF[13698] channel.c: DTMF end accepted with begin '3' on Recorder/ARI-0000001e;1 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DTMF[13698] channel.c: DTMF end passthrough '3' on Recorder/ARI-0000001e;1 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20535] stasis.c: Topic 'devicestate:all/Snoop/212993': 0x7f0c84109620 created [Aug 18 10:34:00] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #51 (2) BYE - 8 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #51)) [Aug 18 10:34:00] DEBUG[13759] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13756] http.c: Match made with [ari] [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20616] app_queue.c: Device 'Snoop/212993' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 From: ;tag=as3edf3f1c To: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Finding handler for bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a/addChannel [Aug 18 10:34:00] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:34:00] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 944, ms is 79 [Aug 18 10:34:00] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 2096, ms is 151 [Aug 18 10:34:00] DEBUG[13758] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:00] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:34:00] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:34:00] DEBUG[13711] channel.c: Channel 0x7f0c0806c2f0 'Announcer/ARI-00000022;2' allocated [Aug 18 10:34:00] DEBUG[13711] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:00] DEBUG[13711] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000022;1' [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Finding handler for e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13757] res_ari.c: No explicit handler found for e0573cd4-75f6-4425-a1e4-83029f01aa9a. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Finding handler for addChannel [Aug 18 10:34:00] DEBUG[13757] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:00] DEBUG[13757] stasis/control.c: 1629282839.182: Sending channel add_to_bridge command [Aug 18 10:34:00] DEBUG[13758] http.c: match request [ari/bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[13747] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:00] DEBUG[13747] chan_sip.c: SIP call-id changed from '7f953ed1633f6daa4c05d2ee361a4aa8@127.0.1.1:5060' to '1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060' [Aug 18 10:34:00] DEBUG[13747] stasis.c: Creating topic. name: channel:213061, detail: [Aug 18 10:34:00] DEBUG[13747] stasis.c: Topic 'channel:213061': 0x7f0c8c00eb50 created [Aug 18 10:34:00] DEBUG[13747] stasis.c: Creating topic. name: cache:244/channel:213061, detail: [Aug 18 10:34:00] DEBUG[13747] stasis.c: Topic 'cache:244/channel:213061': 0x7f0c8c00eca0 created [Aug 18 10:34:00] DEBUG[13760] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pushing 0x7f0c90058340(Announcer/ARI-00000021;2) [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Finding handler for bridges/8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:34:00] DEBUG[13761] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c0804a470(Announcer/ARI-00000022;2) is joining [Aug 18 10:34:00] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:00] DEBUG[13758] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 656, ms is 61 [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Finding handler for 8b092052-108a-4921-8aad-1aecb4e2c824 [Aug 18 10:34:00] DEBUG[13759] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13759] res_ari.c: No explicit handler found for 8b092052-108a-4921-8aad-1aecb4e2c824. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13759] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: telling all channels to leave the party [Aug 18 10:34:00] DEBUG[13759] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:00] DEBUG[13759] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: queueing action type:13 sub:1001 [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:00] DEBUG[13759] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: calling stasis bridge destructor [Aug 18 10:34:00] DEBUG[13686] bridge_roles.c: Roles did not exist on channel Snoop/213008-0000000a [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 8b092052-108a-4921-8aad-1aecb4e2c824: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13686] stasis/control.c: 1629282839.182: Adding to bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:00] DEBUG[13686] stasis/app.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13760] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:00] DEBUG[13762] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c940356f0(Snoop/213008-0000000a) is joining [Aug 18 10:34:00] DEBUG[13759] http.c: HTTP closing session. Top level [Aug 18 10:34:00] VERBOSE[13760] bridge_channel.c: Channel Announcer/ARI-00000021;2 joined 'simple_bridge' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:00] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '8b092052-108a-4921-8aad-1aecb4e2c824': is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '8b092052-108a-4921-8aad-1aecb4e2c824' unsubscribed from calls_0 [Aug 18 10:34:00] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTCP got report of 100 bytes from 178.62.121.41:16939 [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: cache:81/bridge:8b092052-108a-4921-8aad-1aecb4e2c824, detail: [Aug 18 10:34:00] DEBUG[13564] res_rtp_asterisk.c: (0x7f0c24077280) RTP audio difference is 1312, ms is 102 [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'cache:81/bridge:8b092052-108a-4921-8aad-1aecb4e2c824': 0x7f0c900160a0 destroyed [Aug 18 10:34:00] DEBUG[13762] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: pushing 0x7f0c940356f0(Snoop/213008-0000000a) [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: bridge:8b092052-108a-4921-8aad-1aecb4e2c824, detail: [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'bridge:8b092052-108a-4921-8aad-1aecb4e2c824': 0x7f0c90021250 destroyed [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Finding handler for bridges/a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Finding handler for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59 [Aug 18 10:34:00] DEBUG[13758] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13758] res_ari.c: No explicit handler found for a8c92fa0-b7d8-414e-ab7b-67b9dd298b59. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13758] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: telling all channels to leave the party [Aug 18 10:34:00] DEBUG[13758] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:00] DEBUG[13758] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: queueing action type:13 sub:1001 [Aug 18 10:34:00] DEBUG[13760] bridge.c: Chose bridge technology softmix [Aug 18 10:34:00] VERBOSE[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: switching from simple_bridge technology to softmix [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology constructor [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c7803b960(SIP/zvonobot-0000002b) to dummy bridge temporarily [Aug 18 10:34:00] VERBOSE[13762] bridge_channel.c: Channel Snoop/213008-0000000a joined 'simple_bridge' stasis-bridge [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) to dummy bridge temporarily [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is leaving simple_bridge technology (dummy) [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling stasis bridge destructor [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': is 0 interested in calls_0 [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c90058340(Announcer/ARI-00000021;2) is joining softmix technology [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge 'a8c92fa0-b7d8-414e-ab7b-67b9dd298b59' unsubscribed from calls_0 [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Announcer/ARI-00000021;2: [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge a8c92fa0-b7d8-414e-ab7b-67b9dd298b59: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13760] channel.c: Channel Announcer/ARI-00000021;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13766] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Announcer/ARI-00000021;2: [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Announcer/ARI-00000021;2: Not in SFU mode [Aug 18 10:34:00] DEBUG[13766] http.c: HTTP Request URI is /ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining softmix technology [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: SIP/zvonobot-0000002b: [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: cache:94/bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59, detail: [Aug 18 10:34:00] DEBUG[13762] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 From: ;tag=as6ceaa437 To: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag [Aug 18 10:34:00] DEBUG[13762] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13762] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #13 - INVITE (got response) [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'cache:94/bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': 0x7f0c78017da0 destroyed [Aug 18 10:34:00] DEBUG[13762] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13762] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59, detail: [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'bridge:a8c92fa0-b7d8-414e-ab7b-67b9dd298b59': 0x7f0c78007260 destroyed [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13762] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a is already using the new technology. [Aug 18 10:34:00] DEBUG[13766] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2] with handler [httpstatus] len 10 [Aug 18 10:34:00] DEBUG[13761] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pushing 0x7f0c0804a470(Announcer/ARI-00000022;2) [Aug 18 10:34:00] DEBUG[13762] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c940356f0(Snoop/213008-0000000a) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13758] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[13766] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: SIP/zvonobot-0000002b: [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13758] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: SIP/zvonobot-0000002b: Not in SFU mode [Aug 18 10:34:00] DEBUG[13766] http.c: match request [ari/bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2] with handler [ari] len 3 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 From: ;tag=as686a751a To: ;tag=as6e8cb5a3 Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1721442823 1721442823 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11280 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f49e3d4;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as686a751a [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e8cb5a3 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1721442823 1721442823 IN IP4 178.62.121.41 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11280 RTP/AVP 0 8 101 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining softmix technology [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[13766] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 (Checking To) --From tag as686a751a --To-tag as6e8cb5a3 [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Recorder/ARI-0000001a;2: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Stopping retransmission on '398559732fb8625271bea90231b90490@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Strict routing enforced for session 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:00] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:00] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117033@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK723047e9 Max-Forwards: 70 From: ;tag=as686a751a To: ;tag=as6e8cb5a3 Contact: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:00] DEBUG[13720] channel.c: Channel 0x7f0c2c0a7290 'SIP/zvonobot-0000005a' allocated [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:00] DEBUG[13720] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:00] DEBUG[13686] stasis/app.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' is 2 interested in calls_0 [Aug 18 10:34:00] DEBUG[13757] stasis/control.c: robot_213008: Sending channel add_to_bridge command [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13720] res_stasis.c: calls_0: Subscribing to 213056 [Aug 18 10:34:00] DEBUG[13720] stasis/app.c: Channel '213056' is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13760] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13760] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Recorder/ARI-0000001a;2: [Aug 18 10:34:00] DEBUG[13760] bridge_softmix.c: Recorder/ARI-0000001a;2: Not in SFU mode [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology start [Aug 18 10:34:00] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Finding handler for bridges/9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:34:00] DEBUG[13720] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:00] DEBUG[13720] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Finding handler for bridges [Aug 18 10:34:00] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Outgoing Call for 79821116984 [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:00] DEBUG[13756] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13739] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:00] DEBUG[13739] stasis/control.c: robot_213008: Adding to bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:00] DEBUG[13739] stasis/app.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' is 3 interested in calls_0 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:00] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 640, ms is 60 [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Finding handler for 9d1bf1e2-893f-4249-b006-4b3a345e76a2 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:00] DEBUG[13768] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) is joining [Aug 18 10:34:00] DEBUG[13766] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:00] DEBUG[13766] res_ari.c: No explicit handler found for 9d1bf1e2-893f-4249-b006-4b3a345e76a2. Using wildcard bridgeId. [Aug 18 10:34:00] DEBUG[13764] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: starting mixing thread [Aug 18 10:34:00] DEBUG[13691] res_stasis_playback.c: 1629282839.188: Sending play(sound:silence/2) command [Aug 18 10:34:00] DEBUG[13766] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: telling all channels to leave the party [Aug 18 10:34:00] DEBUG[13768] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: pushing 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:00] DEBUG[13766] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:00] DEBUG[13691] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:00] DEBUG[13766] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: queueing action type:13 sub:1001 [Aug 18 10:34:00] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:00] DEBUG[13691] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13766] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:00] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:00] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP ooh, format changed from none to ulaw [Aug 18 10:34:00] DEBUG[13766] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13756] stasis.c: Creating topic. name: bridge:28c87384-44a9-4ebc-9328-4118df068e33, detail: [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: calling stasis bridge destructor [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:00] DEBUG[20534] bridge.c: Bridge 9d1bf1e2-893f-4249-b006-4b3a345e76a2: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2': is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20620] stasis/app.c: bridge '9d1bf1e2-893f-4249-b006-4b3a345e76a2' unsubscribed from calls_0 [Aug 18 10:34:00] DEBUG[13770] channel.c: Channel Announcer/ARI-00000021;1 setting write format path: gsm -> slin [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] VERBOSE[13768] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13756] stasis.c: Topic 'bridge:28c87384-44a9-4ebc-9328-4118df068e33': 0x7f0ca00777e0 created [Aug 18 10:34:00] DEBUG[13756] stasis.c: Creating topic. name: cache:245/bridge:28c87384-44a9-4ebc-9328-4118df068e33, detail: [Aug 18 10:34:00] DEBUG[13756] stasis.c: Topic 'cache:245/bridge:28c87384-44a9-4ebc-9328-4118df068e33': 0x7f0ca002e970 created [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: cache:99/bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2, detail: [Aug 18 10:34:00] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 - start 1629282840.133076 answer 1629282840.192908 end 1629282840.855650 dur 0.722 bill 0.662 dispo ANSWERED [Aug 18 10:34:00] DEBUG[13723] channel.c: Channel 0x7f0c280d3fe0 'SIP/zvonobot-0000005b' allocated [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:00] DEBUG[13723] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'cache:99/bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2': 0x7f0cb008d500 destroyed [Aug 18 10:34:00] VERBOSE[13767] chan_sip.c: Audio is at 19866 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag [Aug 18 10:34:00] DEBUG[13768] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a'. Checking compatability for channels 'Snoop/213008-0000000a' and 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40' [Aug 18 10:34:00] DEBUG[13768] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' can not use native RTP bridge as could not get details [Aug 18 10:34:00] DEBUG[13761] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:00] DEBUG[20534] stasis.c: Destroying topic. name: bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2, detail: [Aug 18 10:34:00] DEBUG[13768] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[20534] stasis.c: Topic 'bridge:9d1bf1e2-893f-4249-b006-4b3a345e76a2': 0x7f0cb007f410 destroyed [Aug 18 10:34:00] DEBUG[13768] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] VERBOSE[13767] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:00] DEBUG[13756] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' can not use native RTP bridge as two channels are required [Aug 18 10:34:00] DEBUG[13756] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:00] DEBUG[13756] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13756] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:00] DEBUG[13756] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13756] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: calling simple_bridge technology constructor [Aug 18 10:34:00] DEBUG[13756] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: calling simple_bridge technology start [Aug 18 10:34:00] DEBUG[13756] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:00] DEBUG[13768] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13770] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:00] VERBOSE[13770] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:00] DEBUG[13771] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[13771] http.c: HTTP Request URI is /ari/channels/213011/snoop?app=calls_0&spy=in [Aug 18 10:34:00] DEBUG[13768] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:00] DEBUG[13768] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a is already using the new technology. [Aug 18 10:34:00] VERBOSE[13767] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:00] DEBUG[13756] http.c: HTTP closing session. Top level [Aug 18 10:34:00] VERBOSE[13767] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Initializing initreq for method INVITE - callid 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116984@178.62.121.41 SIP/2.0 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 3 [ 52]: From: ;tag=as000dc064 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 6 [ 60]: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:00 GMT [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:00] VERBOSE[13767] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[13768] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) is joining simple_bridge technology [Aug 18 10:34:00] DEBUG[13771] http.c: match request [ari/channels/213011/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:00] VERBOSE[13761] bridge_channel.c: Channel Announcer/ARI-00000022;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:00] DEBUG[13768] channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 setting read format path: slin16 -> slin16 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Aug 18 10:34:00] DEBUG[13771] http.c: match request [ari/channels/213011/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13767] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13771] http.c: match request [ari/channels/213011/snoop] with handler [ari] len 3 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[13768] channel.c: Channel Snoop/213008-0000000a setting write format path: slin16 -> slin [Aug 18 10:34:00] DEBUG[13771] http.c: Match made with [ari] [Aug 18 10:34:00] DEBUG[13723] res_stasis.c: calls_0: Subscribing to 213054 [Aug 18 10:34:00] DEBUG[13723] stasis/app.c: Channel '213054' is 1 interested in calls_0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:00] DEBUG[13723] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:00] DEBUG[13723] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13768] channel.c: Channel Snoop/213008-0000000a setting read format path: slin -> slin16 [Aug 18 10:34:00] DEBUG[13768] channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 setting write format path: slin16 -> slin16 [Aug 18 10:34:00] DEBUG[13771] res_ari.c: Finding handler for channels/213011/snoop [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[13771] res_ari.c: Finding handler for channels [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce;received=159.65.48.104 From: ;tag=as52d5bd88 To: ;tag=as171b84c8 Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1cff09" Content-Length: 0 <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ea3c9ce;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as171b84c8 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[13739] stasis/app.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' is 4 interested in calls_0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1cff09" [Aug 18 10:34:00] DEBUG[13757] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:00] DEBUG[13757] http.c: HTTP closing session. Top level [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:00] DEBUG[13761] bridge.c: Chose bridge technology softmix [Aug 18 10:34:00] VERBOSE[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: switching from simple_bridge technology to softmix [Aug 18 10:34:00] DEBUG[13771] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology constructor [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c9809c220(SIP/zvonobot-0000000e) to dummy bridge temporarily [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) to dummy bridge temporarily [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is leaving simple_bridge technology (dummy) [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology stop [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c0804a470(Announcer/ARI-00000022;2) is joining softmix technology [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Announcer/ARI-00000022;2: [Aug 18 10:34:00] DEBUG[13761] channel.c: Channel Announcer/ARI-00000022;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Announcer/ARI-00000022;2: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Announcer/ARI-00000022;2: Not in SFU mode [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is joining softmix technology [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: SIP/zvonobot-0000000e: [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: SIP/zvonobot-0000000e: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: SIP/zvonobot-0000000e: Not in SFU mode [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining softmix technology [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Recorder/ARI-00000019;2: [Aug 18 10:34:00] DEBUG[13761] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:00] DEBUG[13761] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Recorder/ARI-00000019;2: [Aug 18 10:34:00] DEBUG[13761] bridge_softmix.c: Recorder/ARI-00000019;2: Not in SFU mode [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology start [Aug 18 10:34:00] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology destructor [Aug 18 10:34:00] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 720, ms is 65 [Aug 18 10:34:00] VERBOSE[13767] dial.c: Called zvonobot/79821116984 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag as171b84c8 [Aug 18 10:34:00] DEBUG[13771] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:00] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:00] DEBUG[13774] http.c: HTTP opening session. Top level [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Stopping retransmission on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:00] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:00] DEBUG[13382] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[13774] http.c: HTTP Request URI is /ari/playbacks/1146fe80-2b3e-4a47-9336-9e99c15c6b31 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #51 (3) BYE - 8 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #51)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #64 (4) INVITE - 5 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #64)) [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 From: ;tag=as73737e94 To: ;tag=as6e22f1d1 Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 870064292 870064292 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16540 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6bc2dcfa;received=159.65.48.104 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as73737e94 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e22f1d1 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 870064292 870064292 IN IP4 178.62.121.41 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16540 RTP/AVP 0 8 101 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:00] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:00] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 (Checking To) --From tag as73737e94 --To-tag as6e22f1d1 [Aug 18 10:34:00] DEBUG[13714] channel.c: Channel 0x7f0c1c120cb0 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20' allocated [Aug 18 10:34:00] DEBUG[13714] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:00] VERBOSE[13714] res_rtp_asterisk.c: 0x7f0c1c0b4640 -- Strict RTP learning after remote address set to: 127.0.0.1:50194 [Aug 18 10:34:00] DEBUG[13772] chan_sip.c: Outgoing Call for 79821116986 [Aug 18 10:34:00] DEBUG[13771] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Stopping retransmission on '79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:01] DEBUG[13714] res_stasis.c: calls_0: Subscribing to robot_212977 [Aug 18 10:34:01] DEBUG[13714] stasis/app.c: Channel 'robot_212977' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13768] channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:01] DEBUG[13768] channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 setting write format path: slin -> slin16 [Aug 18 10:34:01] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP ooh, format changed from none to slin16 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Strict routing enforced for session 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:01] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:01] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117031@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7df47b24 Max-Forwards: 70 From: ;tag=as73737e94 To: ;tag=as6e22f1d1 Contact: Call-ID: 79ebefe12b9e09421a4a34432fab1705@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #14 (1) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #14)) [Aug 18 10:34:01] DEBUG[13774] http.c: match request [ari/playbacks/1146fe80-2b3e-4a47-9336-9e99c15c6b31] with handler [httpstatus] len 10 [Aug 18 10:34:01] DEBUG[13774] http.c: match request [ari/playbacks/1146fe80-2b3e-4a47-9336-9e99c15c6b31] with handler [phoneprov] len 9 [Aug 18 10:34:00] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13714] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13714] http.c: HTTP closing session. Top level [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag [Aug 18 10:34:01] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:00] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Finding handler for 213011 [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channels create: Didn't match 213011 [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channels externalMedia: Didn't match 213011 [Aug 18 10:34:01] DEBUG[13771] res_ari.c: No explicit handler found for 213011. Using wildcard channelId. [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Finding handler for snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:01] DEBUG[13771] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:01] DEBUG[13774] http.c: match request [ari/playbacks/1146fe80-2b3e-4a47-9336-9e99c15c6b31] with handler [ari] len 3 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #48 - INVITE (got response) [Aug 18 10:34:01] DEBUG[13774] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 From: ;tag=as6ceaa437 To: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag [Aug 18 10:34:01] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 760, ms is 115 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (4) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Finding handler for playbacks/1146fe80-2b3e-4a47-9336-9e99c15c6b31 [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Finding handler for playbacks [Aug 18 10:34:01] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13711] res_stasis_playback.c: 1629282839.189: Sending play(sound:silence/2) command [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13711] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[13711] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] VERBOSE[13772] chan_sip.c: Audio is at 13928 [Aug 18 10:34:01] VERBOSE[13772] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] VERBOSE[13772] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] VERBOSE[13772] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Finding handler for 1146fe80-2b3e-4a47-9336-9e99c15c6b31 [Aug 18 10:34:01] DEBUG[13774] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13774] res_ari.c: No explicit handler found for 1146fe80-2b3e-4a47-9336-9e99c15c6b31. Using wildcard playbackId. [Aug 18 10:34:01] DEBUG[13774] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:01] DEBUG[13774] http.c: HTTP closing session. Top level [Aug 18 10:34:01] VERBOSE[13777] dial.c: Called 127.0.0.1:50194 [Aug 18 10:34:01] DEBUG[13509] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:01] DEBUG[13509] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:01] DEBUG[13509] channel.c: Channel Announcer/ARI-00000016;1 setting write format path: slin -> slin [Aug 18 10:34:01] NOTICE[13509] res_stasis_playback.c: 1629282835.136: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:01] DEBUG[13509] channel.c: Channel 0x7f0c2005c6f0 'Announcer/ARI-00000016;1' hanging up. Refs: 2 [Aug 18 10:34:01] DEBUG[13454] res_rtp_asterisk.c: (0x7f0c9800afc0) RTP audio difference is 960, ms is 140 [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Initializing initreq for method INVITE - callid 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116986@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13775] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: starting mixing thread [Aug 18 10:34:01] DEBUG[13780] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 3 [ 52]: From: ;tag=as3da39e97 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 6 [ 60]: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13772] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Aug 18 10:34:01] DEBUG[13772] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:01] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP ooh, format changed from none to ulaw [Aug 18 10:34:01] VERBOSE[13772] dial.c: Called zvonobot/79821116986 [Aug 18 10:34:01] VERBOSE[13777] dial.c: UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 answered [Aug 18 10:34:01] VERBOSE[13777] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[13780] http.c: HTTP Request URI is /ari/channels/robot_212973 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:01] DEBUG[13780] http.c: match request [ari/channels/robot_212973] with handler [httpstatus] len 10 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 From: ;tag=as2b432b30 To: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[13780] http.c: match request [ari/channels/robot_212973] with handler [phoneprov] len 9 [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13780] http.c: match request [ari/channels/robot_212973] with handler [ari] len 3 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13738] channel.c: Channel 0x7f0c4006f090 'SIP/zvonobot-0000005c' allocated [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13738] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[13780] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Finding handler for channels/robot_212973 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13738] res_stasis.c: calls_0: Subscribing to 213059 [Aug 18 10:34:01] DEBUG[13738] stasis/app.c: Channel '213059' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Finding handler for channels [Aug 18 10:34:01] DEBUG[13738] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13778] channel.c: Channel Announcer/ARI-00000022;1 setting write format path: gsm -> slin [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag [Aug 18 10:34:01] DEBUG[13738] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13777] stasis/app.c: Channel 'robot_212977' is 2 interested in calls_0 [Aug 18 10:34:01] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Outgoing Call for 79821116981 [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:01] DEBUG[13784] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13778] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:01] VERBOSE[13556] res_rtp_asterisk.c: 0x7f0c7c0228e0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:14674 [Aug 18 10:34:01] VERBOSE[13778] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:01] DEBUG[13784] http.c: HTTP Request URI is /ari/bridges/48086187-3f40-424c-b978-0d6c6da7141b/addChannel?channel=1629282839.183%2Crobot_212977 [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[13784] http.c: match request [ari/bridges/48086187-3f40-424c-b978-0d6c6da7141b/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13784] http.c: match request [ari/bridges/48086187-3f40-424c-b978-0d6c6da7141b/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:01] DEBUG[13784] http.c: match request [ari/bridges/48086187-3f40-424c-b978-0d6c6da7141b/addChannel] with handler [ari] len 3 [Aug 18 10:34:01] DEBUG[13750] channel.c: Channel 0x7f0c9407a670 'Recorder/ARI-00000024;1' allocated [Aug 18 10:34:01] DEBUG[13750] stasis.c: Creating topic. name: channel:1629282841.207, detail: [Aug 18 10:34:01] DEBUG[13750] stasis.c: Topic 'channel:1629282841.207': 0x7f0c94057d20 created [Aug 18 10:34:01] DEBUG[13750] stasis.c: Creating topic. name: cache:246/channel:1629282841.207, detail: [Aug 18 10:34:01] DEBUG[13750] stasis.c: Topic 'cache:246/channel:1629282841.207': 0x7f0c94063c20 created [Aug 18 10:34:01] DEBUG[13238] channel.c: Channel 0x7f0c34024da0 'SIP/zvonobot-00000016' destroying [Aug 18 10:34:01] DEBUG[13736] channel.c: Channel 0x7f0c38056500 'SIP/zvonobot-0000005d' allocated [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:01] DEBUG[13716] channel.c: Channel 0x7f0c180b81a0 'Snoop/213023-0000000c' allocated [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:01] DEBUG[13784] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[13716] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13716] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13736] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[13703] bridge_channel.c: Setting 0x7f0c240f8830(Announcer/ARI-0000001f;2) state from:0 to:1 [Aug 18 10:34:01] DEBUG[13732] channel.c: Channel 0x7f0c240f5d70 'Announcer/ARI-0000001f;1' destroying [Aug 18 10:34:01] DEBUG[13703] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pulling 0x7f0c240f8830(Announcer/ARI-0000001f;2) [Aug 18 10:34:01] VERBOSE[13703] bridge_channel.c: Channel Announcer/ARI-0000001f;2 left 'softmix' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:34:01] DEBUG[13703] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c240f8830(Announcer/ARI-0000001f;2) is leaving softmix technology [Aug 18 10:34:01] DEBUG[13790] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13627] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee'. Checking compatability for channels 'SIP/zvonobot-0000003b' and 'Recorder/ARI-0000001e;2' [Aug 18 10:34:01] DEBUG[13627] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as channel 'SIP/zvonobot-0000003b' has features which prevent it [Aug 18 10:34:01] DEBUG[13627] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13627] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13627] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13627] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[13627] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee is already using the new technology. [Aug 18 10:34:01] DEBUG[13790] http.c: HTTP Request URI is /ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play?media=sound%3Asilence%2F2 [Aug 18 10:34:01] DEBUG[13238] channel.c: Channel 0x7f0ca8007d50 'Snoop/212986-00000002' destroying [Aug 18 10:34:01] DEBUG[13238] stasis.c: Destroying topic. name: cache:82/channel:1629282831.69, detail: [Aug 18 10:34:01] DEBUG[13238] stasis.c: Topic 'cache:82/channel:1629282831.69': 0x7f0ca800ea90 destroyed [Aug 18 10:34:01] DEBUG[13238] stasis.c: Destroying topic. name: channel:1629282831.69, detail: [Aug 18 10:34:01] DEBUG[13238] stasis.c: Topic 'channel:1629282831.69': 0x7f0ca802e8b0 destroyed [Aug 18 10:34:01] DEBUG[13732] stasis.c: Destroying topic. name: cache:213/channel:1629282839.179, detail: [Aug 18 10:34:01] DEBUG[13732] stasis.c: Topic 'cache:213/channel:1629282839.179': 0x7f0c240f8a40 destroyed [Aug 18 10:34:01] DEBUG[13732] stasis.c: Destroying topic. name: channel:1629282839.179, detail: [Aug 18 10:34:01] DEBUG[13732] stasis.c: Topic 'channel:1629282839.179': 0x7f0c240facd0 destroyed [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282841.208, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'channel:1629282841.208': 0x7f0c300ba000 created [Aug 18 10:34:01] DEBUG[20545] stasis.c: Creating topic. name: cache:247/channel:1629282841.208, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'cache:247/channel:1629282841.208': 0x7f0c300593a0 created [Aug 18 10:34:01] DEBUG[13736] res_stasis.c: calls_0: Subscribing to 213057 [Aug 18 10:34:01] DEBUG[13736] stasis/app.c: Channel '213057' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Finding handler for bridges/48086187-3f40-424c-b978-0d6c6da7141b/addChannel [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Finding handler for bridges [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:01] DEBUG[13703] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162'. Checking compatability for channels 'SIP/zvonobot-00000023' and 'Recorder/ARI-00000013;2' [Aug 18 10:34:01] DEBUG[13703] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as channel 'SIP/zvonobot-00000023' has features which prevent it [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13703] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] VERBOSE[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: switching from softmix technology to simple_bridge [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology constructor [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0ca0053060(SIP/zvonobot-00000023) to dummy bridge temporarily [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: moving 0x7f0c7c018d60(Recorder/ARI-00000013;2) to dummy bridge temporarily [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is leaving softmix technology (dummy) [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is leaving softmix technology (dummy) [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology stop [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is joining simple_bridge technology [Aug 18 10:34:01] DEBUG[13703] channel.c: Channel Recorder/ARI-00000013;2 setting read format path: slin -> slin [Aug 18 10:34:01] DEBUG[13703] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is joining simple_bridge technology [Aug 18 10:34:01] DEBUG[13736] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13703] channel.c: Channel Recorder/ARI-00000013;2 setting read format path: slin -> slin [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Outgoing Call for 79821116983 [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:01] DEBUG[13736] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Finding handler for robot_212973 [Aug 18 10:34:01] VERBOSE[13782] chan_sip.c: Audio is at 11858 [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking channels create: Didn't match robot_212973 [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13780] res_ari.c: Checking channels externalMedia: Didn't match robot_212973 [Aug 18 10:34:01] DEBUG[13780] res_ari.c: No explicit handler found for robot_212973. Using wildcard channelId. [Aug 18 10:34:01] DEBUG[13792] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] VERBOSE[13782] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] VERBOSE[13782] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] VERBOSE[13782] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:01] DEBUG[20620] stasis/app.c: channel '212986': is 0 interested in calls_0 [Aug 18 10:34:01] DEBUG[20620] stasis/app.c: channel '212986' unsubscribed from calls_0 [Aug 18 10:34:01] DEBUG[20620] stasis.c: Destroying topic. name: cache:29/channel:212986, detail: [Aug 18 10:34:01] DEBUG[20620] stasis.c: Topic 'cache:29/channel:212986': 0x7f0c34027950 destroyed [Aug 18 10:34:01] DEBUG[20620] stasis.c: Destroying topic. name: channel:212986, detail: [Aug 18 10:34:01] DEBUG[20620] stasis.c: Topic 'channel:212986': 0x7f0c34026ee0 destroyed [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:01] DEBUG[13792] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213023&app=calls_0&format=slin16&external_host=127.0.0.1%3A50430 [Aug 18 10:34:01] DEBUG[20545] stasis.c: Destroying topic. name: cache:247/channel:1629282841.208, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'cache:247/channel:1629282841.208': 0x7f0c300593a0 destroyed [Aug 18 10:34:01] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282841.208, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'channel:1629282841.208': 0x7f0c300ba000 destroyed [Aug 18 10:34:01] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000016', '', 'Stasis', 'calls_0', 13, 8, 'ANSWERED', 3, '', '212986', '')] [Aug 18 10:34:01] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Finding handler for 48086187-3f40-424c-b978-0d6c6da7141b [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[13792] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:01] DEBUG[13703] channel.c: Channel Recorder/ARI-00000013;2 setting write format path: slin -> slin [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology start [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: deferring softmix technology destructor [Aug 18 10:34:01] DEBUG[13703] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: queueing action type:13 sub:1000 [Aug 18 10:34:01] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 688, ms is 63 [Aug 18 10:34:01] DEBUG[13784] res_ari.c: No explicit handler found for 48086187-3f40-424c-b978-0d6c6da7141b. Using wildcard bridgeId. [Aug 18 10:34:01] DEBUG[13792] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:01] DEBUG[13790] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [httpstatus] len 10 [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Finding handler for addChannel [Aug 18 10:34:01] DEBUG[13784] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:01] DEBUG[13792] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:01] VERBOSE[13550] res_rtp_asterisk.c: 0x7f0c9c00a550 -- Strict RTP learning complete - Locking on source address 178.62.121.41:15418 [Aug 18 10:34:01] DEBUG[13790] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [phoneprov] len 9 [Aug 18 10:34:01] DEBUG[13784] stasis/control.c: 1629282839.183: Sending channel add_to_bridge command [Aug 18 10:34:01] DEBUG[13790] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [ari] len 3 [Aug 18 10:34:01] DEBUG[13790] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Finding handler for bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Finding handler for bridges [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282841.209, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'channel:1629282841.209': 0x7f0c300ba000 created [Aug 18 10:34:01] DEBUG[20545] stasis.c: Creating topic. name: cache:248/channel:1629282841.209, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'cache:248/channel:1629282841.209': 0x7f0c300fde90 created [Aug 18 10:34:01] DEBUG[20545] stasis.c: Destroying topic. name: cache:248/channel:1629282841.209, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'cache:248/channel:1629282841.209': 0x7f0c300fde90 destroyed [Aug 18 10:34:01] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282841.209, detail: [Aug 18 10:34:01] DEBUG[20545] stasis.c: Topic 'channel:1629282841.209': 0x7f0c300ba000 destroyed [Aug 18 10:34:01] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:51', '"" <>', '', 's', 'default', 'Snoop/212986-00000002', 'UnicastRTP/127.0.0.1:50291-0x7f0ca0023720', 'Stasis', 'calls_0', 8, 8, 'ANSWERED', 3, '', '1629282831.69', '')] [Aug 18 10:34:01] DEBUG[13706] bridge_roles.c: Roles did not exist on channel Snoop/212977-0000000b [Aug 18 10:34:01] DEBUG[13706] stasis/control.c: 1629282839.183: Adding to bridge 48086187-3f40-424c-b978-0d6c6da7141b [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] DEBUG[13792] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[13785] stasis/app.c: Channel '1629282840.199' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13706] stasis/app.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:01] DEBUG[13796] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c100f0220(Snoop/212977-0000000b) is joining [Aug 18 10:34:01] DEBUG[13715] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: stopping mixing thread [Aug 18 10:34:01] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:01] VERBOSE[13793] chan_sip.c: Audio is at 13840 [Aug 18 10:34:01] VERBOSE[13793] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] VERBOSE[13793] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] VERBOSE[13793] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[20534] bridge_softmix.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: Waiting for mixing thread to die. [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Finding handler for channels [Aug 18 10:34:01] DEBUG[13462] channel.c: Recorder/ARI-00000013;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:01] DEBUG[13454] channel.c: SIP/zvonobot-00000023: Dropping redundant connected line update "" <>. [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Initializing initreq for method INVITE - callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 From: ;tag=as3edf3f1c To: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Finding handler for 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:34:01] DEBUG[13785] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 46 instead [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Initializing initreq for method INVITE - callid 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116983@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 3 [ 52]: From: ;tag=as00c25c39 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 6 [ 60]: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116981@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 3 [ 52]: From: ;tag=as76ba9cfd [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 6 [ 60]: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13782] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #48 [Aug 18 10:34:01] DEBUG[13782] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13792] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:01] DEBUG[13796] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: pushing 0x7f0c100f0220(Snoop/212977-0000000b) [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:01] VERBOSE[13782] dial.c: Called zvonobot/79821116981 [Aug 18 10:34:01] DEBUG[13703] channel.c: Channel 0x7f0c24016f20 'Announcer/ARI-0000001f;2' hanging up. Refs: 2 [Aug 18 10:34:01] DEBUG[13792] netsock2.c: Splitting '127.0.0.1:50430' into... [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13792] netsock2.c: ...host '127.0.0.1' and port '50430'. [Aug 18 10:34:01] DEBUG[13790] res_ari.c: No explicit handler found for 357a4882-a24d-489f-8ff8-98badd81b2ee. Using wildcard bridgeId. [Aug 18 10:34:01] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13785] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] VERBOSE[13796] bridge_channel.c: Channel Snoop/212977-0000000b joined 'simple_bridge' stasis-bridge <48086187-3f40-424c-b978-0d6c6da7141b> [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[13777] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Finding handler for play [Aug 18 10:34:01] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP creating BEGIN DTMF Frame: 51 (3), at 178.62.121.41:16540 [Aug 18 10:34:01] VERBOSE[13793] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF begin '3' received on SIP/zvonobot-0000002f [Aug 18 10:34:01] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF begin passthrough '3' on SIP/zvonobot-0000002f [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:34:01] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:01] DEBUG[13792] netsock2.c: Splitting '127.0.0.1:50430' into... [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13796] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' can not use native RTP bridge as two channels are required [Aug 18 10:34:01] DEBUG[13796] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13796] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13796] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13796] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[13796] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b is already using the new technology. [Aug 18 10:34:01] DEBUG[13796] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c100f0220(Snoop/212977-0000000b) is joining simple_bridge technology [Aug 18 10:34:01] DEBUG[13792] netsock2.c: ...host '127.0.0.1' and port '50430'. [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:01] DEBUG[13792] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17 [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:01] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:01] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13792] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7804a920' [Aug 18 10:34:01] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP creating END DTMF Frame: 51 (3), at 178.62.121.41:16540 [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF end '3' received on SIP/zvonobot-0000002f, duration 100 ms [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) RTP allocated port 14222 [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF end accepted with begin '3' on SIP/zvonobot-0000002f [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF end '3' detected to have actual duration 44 on the wire, emulation will be triggered on SIP/zvonobot-0000002f [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF end '3' has duration 44 but want minimum 80, emulating on SIP/zvonobot-0000002f [Aug 18 10:34:01] DEBUG[13793] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:01] DEBUG[13784] stasis/control.c: robot_212977: Sending channel add_to_bridge command [Aug 18 10:34:01] DEBUG[13706] stasis/app.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' is 2 interested in calls_0 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 505 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 505 [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) ICE creating session 127.0.0.1:14222 (14222) [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) ICE create [Aug 18 10:34:01] DEBUG[13790] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:01] DTMF[13695] channel.c: DTMF end emulation of '3' queued on SIP/zvonobot-0000002f [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (1) INVITE - 5 [Aug 18 10:34:01] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13790] stasis.c: Creating topic. name: channel:1629282841.210, detail: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:01] DEBUG[13790] stasis.c: Topic 'channel:1629282841.210': 0x7f0c8408a4f0 created [Aug 18 10:34:01] DEBUG[13790] stasis.c: Creating topic. name: cache:249/channel:1629282841.210, detail: [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13790] stasis.c: Topic 'cache:249/channel:1629282841.210': 0x7f0c84101300 created [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #14 (2) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #14)) [Aug 18 10:34:01] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) ICE add system candidates [Aug 18 10:34:01] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:01] VERBOSE[13793] dial.c: Called zvonobot/79821116983 [Aug 18 10:34:01] DEBUG[12903] chan_sip.c: Hangup call SIP/zvonobot-00000009, SIP callid 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[12903] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:01] DEBUG[12903] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:01] DEBUG[13792] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:01] DEBUG[13728] channel.c: Channel 0x7f0c340fb900 'SIP/zvonobot-0000005e' allocated [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13728] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[13728] res_stasis.c: calls_0: Subscribing to 213055 [Aug 18 10:34:01] DEBUG[13728] stasis/app.c: Channel '213055' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13728] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13728] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13717] channel.c: Channel 0x7f0c20090f90 'Recorder/ARI-00000023;2' allocated [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13382] bridge_channel.c: Setting 0x7f0c70055800(Snoop/212973-00000005) state from:0 to:1 [Aug 18 10:34:01] DEBUG[13792] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:01] DEBUG[13717] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Outgoing Call for 79821116985 [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) ICE add candidate: 159.65.48.104:14222, 2130706431 [Aug 18 10:34:01] DEBUG[13792] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:01] DEBUG[13792] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:01] DEBUG[13792] res_rtp_asterisk.c: (0x7f0c7804a920) ICE add candidate: 10.131.0.10:14222, 2130706431 [Aug 18 10:34:01] DEBUG[13792] rtp_engine.c: RTP instance '0x7f0c7804a920' is setup and ready to go [Aug 18 10:34:01] DEBUG[13792] stasis.c: Creating topic. name: channel:robot_213023, detail: [Aug 18 10:34:01] DEBUG[13792] stasis.c: Topic 'channel:robot_213023': 0x7f0c780728f0 created [Aug 18 10:34:01] DEBUG[13792] stasis.c: Creating topic. name: cache:250/channel:robot_213023, detail: [Aug 18 10:34:01] DEBUG[13792] stasis.c: Topic 'cache:250/channel:robot_213023': 0x7f0c7803b0e0 created [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] VERBOSE[13797] chan_sip.c: Audio is at 14460 [Aug 18 10:34:01] VERBOSE[13797] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] VERBOSE[13797] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] VERBOSE[13797] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13798] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is joining [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[13382] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: pulling 0x7f0c70055800(Snoop/212973-00000005) [Aug 18 10:34:01] VERBOSE[13382] bridge_channel.c: Channel Snoop/212973-00000005 left 'simple_bridge' stasis-bridge <6f6fd705-00c9-4b9f-a75f-d7c933b85e66> [Aug 18 10:34:01] DEBUG[13382] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c70055800(Snoop/212973-00000005) is leaving simple_bridge technology [Aug 18 10:34:01] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212973-00000005 - start 1629282833.568080 answer 1629282833.568080 end 1629282841.524353 dur 7.956 bill 7.956 dispo ANSWERED [Aug 18 10:34:01] DEBUG[13382] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' can not use native RTP bridge as two channels are required [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13382] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13382] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13798] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: pushing 0x7f0c20086d10(Recorder/ARI-00000023;2) [Aug 18 10:34:01] DEBUG[13382] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13382] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13382] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 is already using the new technology. [Aug 18 10:34:01] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 From: ;tag=as574e1b12 To: ;tag=as2c28f3a7 Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 595882522 595882522 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18112 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:01] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK527934ab;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Initializing initreq for method INVITE - callid 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as574e1b12 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2c28f3a7 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 595882522 595882522 IN IP4 178.62.121.41 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116985@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:01] DEBUG[13777] stasis/control.c: robot_212977: Adding to bridge 48086187-3f40-424c-b978-0d6c6da7141b [Aug 18 10:34:01] DEBUG[13777] stasis/app.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' is 3 interested in calls_0 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 3 [ 52]: From: ;tag=as697b28a1 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 6 [ 60]: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13797] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #68 [Aug 18 10:34:01] DEBUG[13797] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18112 RTP/AVP 0 8 101 [Aug 18 10:34:01] DEBUG[13382] bridge_channel.c: Bridge is returning 0x7f0c70055800(Snoop/212973-00000005) to read format slin [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:01] DEBUG[13799] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) is joining [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:01] DEBUG[13382] channel.c: Channel Snoop/212973-00000005 setting read format path: slin -> slin [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[13382] bridge_channel.c: Bridge is returning 0x7f0c70055800(Snoop/212973-00000005) to write format slin [Aug 18 10:34:01] VERBOSE[13797] dial.c: Called zvonobot/79821116985 [Aug 18 10:34:01] DEBUG[13382] channel.c: Channel Snoop/212973-00000005 setting write format path: slin -> slin [Aug 18 10:34:01] DEBUG[13382] stasis/control.c: 1629282833.101, 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: Channel was departed from bridge [Aug 18 10:34:01] DEBUG[13382] stasis/app.c: bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66': is 3 interested in calls_0 [Aug 18 10:34:01] DEBUG[13382] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:01] DEBUG[13360] stasis/control.c: 1629282833.101: Channel departing bridge [Aug 18 10:34:01] DEBUG[13360] bridge.c: Waiting for 0x7f0c70055800(Snoop/212973-00000005) bridge thread to die. [Aug 18 10:34:01] VERBOSE[13619] res_rtp_asterisk.c: 0x7f0c9801eb40 -- Strict RTP learning complete - Locking on source address 178.62.121.41:18792 [Aug 18 10:34:01] DEBUG[13743] channel.c: Channel 0x7f0c78050c90 'SIP/zvonobot-0000005f' allocated [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13743] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[13360] stasis/app.c: channel '1629282833.101': is 0 interested in calls_0 [Aug 18 10:34:01] DEBUG[13360] stasis/app.c: channel '1629282833.101' unsubscribed from calls_0 [Aug 18 10:34:01] DEBUG[13360] channel.c: Channel 0x7f0c40046650 'Snoop/212973-00000005' hanging up. Refs: 3 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:01] DEBUG[13798] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:01] VERBOSE[13798] bridge_channel.c: Channel Recorder/ARI-00000023;2 joined 'simple_bridge' stasis-bridge <0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3> [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:01] DEBUG[13743] res_stasis.c: calls_0: Subscribing to 213058 [Aug 18 10:34:01] DEBUG[13743] stasis/app.c: Channel '213058' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Outgoing Call for 79821116982 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 (Checking To) --From tag as574e1b12 --To-tag as2c28f3a7 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13743] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13743] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13799] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: pushing 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP got report of 100 bytes from 178.62.121.41:14675 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] VERBOSE[13801] chan_sip.c: Audio is at 11586 [Aug 18 10:34:01] VERBOSE[13801] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Stopping retransmission on '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:01] VERBOSE[13801] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:01] VERBOSE[13801] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:01] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP got report of 100 bytes from 178.62.121.41:15419 [Aug 18 10:34:01] VERBOSE[13799] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 joined 'simple_bridge' stasis-bridge <48086187-3f40-424c-b978-0d6c6da7141b> [Aug 18 10:34:01] DEBUG[13798] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3'. Checking compatability for channels 'SIP/zvonobot-0000002f' and 'Recorder/ARI-00000023;2' [Aug 18 10:34:01] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 - start 1629282840.994819 answer 1629282841.087255 end 1629282841.656744 dur 0.661 bill 0.569 dispo ANSWERED [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13798] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' can not use native RTP bridge as could not get details [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117058@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0918616f Max-Forwards: 70 From: ;tag=as574e1b12 To: ;tag=as2c28f3a7 Contact: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:01] DEBUG[13798] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13798] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[13798] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13798] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (1) INVITE - 5 [Aug 18 10:34:01] DEBUG[13798] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 is already using the new technology. [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Initializing initreq for method INVITE - callid 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116982@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (2) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13798] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is joining simple_bridge technology [Aug 18 10:34:01] DEBUG[13798] channel.c: Channel Recorder/ARI-00000023;2 setting read format path: slin -> slin [Aug 18 10:34:01] DEBUG[13798] channel.c: Channel SIP/zvonobot-0000002f setting write format path: slin -> ulaw [Aug 18 10:34:01] DEBUG[13798] channel.c: Channel SIP/zvonobot-0000002f setting read format path: ulaw -> slin [Aug 18 10:34:01] DEBUG[13798] channel.c: Channel Recorder/ARI-00000023;2 setting write format path: slin -> slin [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13799] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b'. Checking compatability for channels 'Snoop/212977-0000000b' and 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20' [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 3 [ 52]: From: ;tag=as08a5ad00 [Aug 18 10:34:01] DEBUG[13799] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' can not use native RTP bridge as could not get details [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (1) INVITE - 5 [Aug 18 10:34:01] DEBUG[13799] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13799] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13799] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13799] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[13799] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b is already using the new technology. [Aug 18 10:34:01] DEBUG[13799] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) is joining simple_bridge technology [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 setting read format path: slin16 -> slin16 [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel Snoop/212977-0000000b setting write format path: slin16 -> slin [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel Snoop/212977-0000000b setting read format path: slin -> slin16 [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 setting write format path: slin16 -> slin16 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 6 [ 60]: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP got report of 76 bytes from 178.62.121.41:18793 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13801] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #70 [Aug 18 10:34:01] DEBUG[13801] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #14 (3) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #14)) [Aug 18 10:34:01] VERBOSE[13801] dial.c: Called zvonobot/79821116982 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13784] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:01] DEBUG[13784] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13777] stasis/app.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' is 4 interested in calls_0 [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13751] channel.c: Channel 0x7f0c9007e3b0 'SIP/zvonobot-00000060' allocated [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13751] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[13503] channel.c: Channel 0x7f0c20061cb0 'Announcer/ARI-00000016;2' destroying [Aug 18 10:34:01] DEBUG[13503] stasis.c: Destroying topic. name: cache:168/channel:1629282835.142, detail: [Aug 18 10:34:01] DEBUG[13503] stasis.c: Topic 'cache:168/channel:1629282835.142': 0x7f0c20040460 destroyed [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13503] stasis.c: Destroying topic. name: channel:1629282835.142, detail: [Aug 18 10:34:01] DEBUG[13503] stasis.c: Topic 'channel:1629282835.142': 0x7f0c2004d4e0 destroyed [Aug 18 10:34:01] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:01] DEBUG[13799] channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 setting write format path: slin -> slin16 [Aug 18 10:34:01] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP ooh, format changed from none to slin16 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (1) INVITE - 5 [Aug 18 10:34:01] DEBUG[13717] res_stasis_recording.c: 1629282840.190: Sending record(213009_KnHWGInDjuQFpdGbHGchCssGOUDQXoGO.wav) command [Aug 18 10:34:01] DEBUG[13717] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:01] DEBUG[13717] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:01] DEBUG[13744] channel.c: Channel 0x7f0c841031a0 'SIP/zvonobot-00000062' allocated [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:01] DEBUG[13744] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:01] DEBUG[13808] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13808] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:01] DEBUG[13807] app.c: play_and_record: , /var/spool/asterisk/recording/213009_KnHWGInDjuQFpdGbHGchCssGOUDQXoGO, 'wav' [Aug 18 10:34:01] DEBUG[13808] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:01] DEBUG[13807] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:01] VERBOSE[13807] app.c: x=0, open writing: /var/spool/asterisk/recording/213009_KnHWGInDjuQFpdGbHGchCssGOUDQXoGO format: wav, 0x7f0c9c06d980 [Aug 18 10:34:01] DEBUG[13808] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13808] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:01] DEBUG[13808] http.c: Match made with [ari] [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Finding handler for bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Finding handler for bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:01] DEBUG[13808] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:01] DEBUG[13808] stasis.c: Creating topic. name: bridge:051b3352-0990-44a6-b6a2-2bd678146686, detail: [Aug 18 10:34:01] DEBUG[13808] stasis.c: Topic 'bridge:051b3352-0990-44a6-b6a2-2bd678146686': 0x7f0c98019ad0 created [Aug 18 10:34:01] DEBUG[13808] stasis.c: Creating topic. name: cache:251/bridge:051b3352-0990-44a6-b6a2-2bd678146686, detail: [Aug 18 10:34:01] DEBUG[13751] res_stasis.c: calls_0: Subscribing to 213062 [Aug 18 10:34:01] DEBUG[13744] res_stasis.c: calls_0: Subscribing to 213060 [Aug 18 10:34:01] DEBUG[13751] stasis/app.c: Channel '213062' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13751] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Outgoing Call for 79821116978 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] VERBOSE[13810] chan_sip.c: Audio is at 17282 [Aug 18 10:34:01] VERBOSE[13810] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] DEBUG[13744] stasis/app.c: Channel '213060' is 1 interested in calls_0 [Aug 18 10:34:01] DEBUG[13808] stasis.c: Topic 'cache:251/bridge:051b3352-0990-44a6-b6a2-2bd678146686': 0x7f0c9809cdf0 created [Aug 18 10:34:01] DEBUG[13751] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13808] bridge_native_rtp.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' can not use native RTP bridge as two channels are required [Aug 18 10:34:01] DEBUG[13808] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:01] DEBUG[13808] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:01] DEBUG[13808] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:01] DEBUG[13808] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:01] DEBUG[13808] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: calling simple_bridge technology constructor [Aug 18 10:34:01] DEBUG[13808] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: calling simple_bridge technology start [Aug 18 10:34:01] DEBUG[13808] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] VERBOSE[13810] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] VERBOSE[13810] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[13744] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Outgoing Call for 79821116980 [Aug 18 10:34:01] DEBUG[13744] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:01] DEBUG[13808] http.c: HTTP closing session. Top level [Aug 18 10:34:01] DEBUG[13812] http.c: HTTP opening session. Top level [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13812] http.c: HTTP Request URI is /ari/channels/213009/snoop?app=calls_0&spy=in [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] DEBUG[13812] http.c: match request [ari/channels/213009/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:01] VERBOSE[13811] chan_sip.c: Audio is at 12964 [Aug 18 10:34:01] VERBOSE[13811] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] DEBUG[13812] http.c: match request [ari/channels/213009/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:01] DEBUG[13812] http.c: match request [ari/channels/213009/snoop] with handler [ari] len 3 [Aug 18 10:34:01] VERBOSE[13811] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] DEBUG[13812] http.c: Match made with [ari] [Aug 18 10:34:01] VERBOSE[13811] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Finding handler for channels/213009/snoop [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Finding handler for channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Finding handler for 213009 [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channels create: Didn't match 213009 [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channels externalMedia: Didn't match 213009 [Aug 18 10:34:01] DEBUG[13812] res_ari.c: No explicit handler found for 213009. Using wildcard channelId. [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Finding handler for snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:01] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:01] DEBUG[13812] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as2aed188c Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" Content-Length: 0 <-------------> [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Initializing initreq for method INVITE - callid 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116978@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 3 [ 52]: From: ;tag=as080d6dff [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 6 [ 60]: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Initializing initreq for method INVITE - callid 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116980@178.62.121.41 SIP/2.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13810] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #72 [Aug 18 10:34:01] DEBUG[13810] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa [Aug 18 10:34:01] VERBOSE[13810] dial.c: Called zvonobot/79821116978 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2aed188c [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as2aed188c [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 3 [ 52]: From: ;tag=as73898a35 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Stopping retransmission on '596ef69a675656833620d8345b47f33c@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: ;tag=as2aed188c Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 6 [ 60]: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:01 GMT [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:01] VERBOSE[13811] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #75 [Aug 18 10:34:01] DEBUG[13811] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:01] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:01] VERBOSE[13811] dial.c: Called zvonobot/79821116980 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:01] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Audio is at 18824 [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:01] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #77 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (2) INVITE - 5 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:01] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:01] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #51 (4) BYE - 8 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #51)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13748] channel.c: Channel 0x7f0c8809bb30 'SIP/zvonobot-00000061' allocated [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:02] DEBUG[13748] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:02] VERBOSE[13356] app.c: User hung up [Aug 18 10:34:02] DEBUG[13356] res_stasis_recording.c: 1629282833.99: Recording complete [Aug 18 10:34:02] DEBUG[13356] channel.c: Channel 0x7f0c240501f0 'Recorder/ARI-0000000d;1' hanging up. Refs: 2 [Aug 18 10:34:02] DEBUG[13353] channel.c: Channel 0x7f0c24056340 'Recorder/ARI-0000000d;2' destroying [Aug 18 10:34:02] DEBUG[13353] stasis.c: Destroying topic. name: cache:119/channel:1629282833.100, detail: [Aug 18 10:34:02] DEBUG[13353] stasis.c: Topic 'cache:119/channel:1629282833.100': 0x7f0c240588f0 destroyed [Aug 18 10:34:02] DEBUG[13353] stasis.c: Destroying topic. name: channel:1629282833.100, detail: [Aug 18 10:34:02] DEBUG[13353] stasis.c: Topic 'channel:1629282833.100': 0x7f0c240517f0 destroyed [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #70 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #70)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 Max-Forwards: 70 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6be15af9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0453a0d2 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking From) --From tag as6be15af9 --To-tag as0453a0d2 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:02] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:02] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Received bye, no owner, selfdestruct soon. [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418;received=178.62.121.41 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982166 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 From: ;tag=as34e313e7 To: ;tag=as51d2bd1a Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as34e313e7 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as51d2bd1a [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 (Checking To) --From tag as34e313e7 --To-tag as51d2bd1a [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #46 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Stopping retransmission on '59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4 Max-Forwards: 70 From: ;tag=as34e313e7 To: ;tag=as51d2bd1a Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Audio is at 11378 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:02] DEBUG[13747] channel.c: Channel 0x7f0c8c0f9790 'SIP/zvonobot-00000063' allocated [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:02] DEBUG[13747] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:02] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:02] DEBUG[13748] res_stasis.c: calls_0: Subscribing to 213063 [Aug 18 10:34:02] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:02] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13748] stasis/app.c: Channel '213063' is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13748] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:02] DEBUG[13748] http.c: HTTP closing session. Top level [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #70 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[13818] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13747] res_stasis.c: calls_0: Subscribing to 213061 [Aug 18 10:34:02] DEBUG[13747] stasis/app.c: Channel '213061' is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #70)) [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13747] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Outgoing Call for 79821116977 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:02] DEBUG[13818] http.c: HTTP Request URI is /ari/channels/213066?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116974&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13747] http.c: HTTP closing session. Top level [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:02] VERBOSE[13819] chan_sip.c: Audio is at 16058 [Aug 18 10:34:02] VERBOSE[13819] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:02] VERBOSE[13819] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:02] VERBOSE[13819] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Initializing initreq for method INVITE - callid 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116977@178.62.121.41 SIP/2.0 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 3 [ 52]: From: ;tag=as6ac21020 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 6 [ 60]: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:02 GMT [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:02] VERBOSE[13819] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13822] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13825] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #65 [Aug 18 10:34:02] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 1120, ms is 90 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Outgoing Call for 79821116979 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:02] DEBUG[13819] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:02] DEBUG[13818] http.c: match request [ari/channels/213066] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13818] http.c: match request [ari/channels/213066] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13818] http.c: match request [ari/channels/213066] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13818] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13818] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Finding handler for channels/213066 [Aug 18 10:34:02] VERBOSE[13820] chan_sip.c: Audio is at 14668 [Aug 18 10:34:02] VERBOSE[13820] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:02] VERBOSE[13820] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:02] VERBOSE[13820] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Initializing initreq for method INVITE - callid 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116979@178.62.121.41 SIP/2.0 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 3 [ 52]: From: ;tag=as54647e8b [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 6 [ 60]: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:02] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:02 GMT [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:02] DEBUG[13822] http.c: HTTP Request URI is /ari/channels/213064?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116976&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13822] http.c: match request [ari/channels/213064] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13832] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13832] http.c: HTTP Request URI is /ari/channels/213065?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116975&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13832] http.c: match request [ari/channels/213065] with handler [httpstatus] len 10 [Aug 18 10:34:02] VERBOSE[13819] dial.c: Called zvonobot/79821116977 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13822] http.c: match request [ari/channels/213064] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13832] http.c: match request [ari/channels/213065] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Finding handler for 213066 [Aug 18 10:34:02] DEBUG[13825] http.c: HTTP Request URI is /ari/channels/213067?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116973&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking channels create: Didn't match 213066 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:02] VERBOSE[13820] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #78 [Aug 18 10:34:02] DEBUG[13820] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13822] http.c: match request [ari/channels/213064] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13818] res_ari.c: Checking channels externalMedia: Didn't match 213066 [Aug 18 10:34:02] DEBUG[13818] res_ari.c: No explicit handler found for 213066. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13822] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13836] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13832] http.c: match request [ari/channels/213065] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13836] http.c: HTTP Request URI is /ari/channels/213068?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116972&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13832] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13822] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13832] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13836] http.c: match request [ari/channels/213068] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13825] http.c: match request [ari/channels/213067] with handler [httpstatus] len 10 [Aug 18 10:34:02] VERBOSE[13820] dial.c: Called zvonobot/79821116979 [Aug 18 10:34:02] DEBUG[13825] http.c: match request [ari/channels/213067] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Finding handler for channels/213064 [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Finding handler for 213064 [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking channels create: Didn't match 213064 [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13822] res_ari.c: Checking channels externalMedia: Didn't match 213064 [Aug 18 10:34:02] DEBUG[13822] res_ari.c: No explicit handler found for 213064. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13836] http.c: match request [ari/channels/213068] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13837] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Finding handler for channels/213065 [Aug 18 10:34:02] DEBUG[13836] http.c: match request [ari/channels/213068] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13746] stasis.c: Creating topic. name: channel:1629282842.212, detail: [Aug 18 10:34:02] DEBUG[13746] stasis.c: Topic 'channel:1629282842.212': 0x7f0c80062df0 created [Aug 18 10:34:02] DEBUG[13746] stasis.c: Creating topic. name: cache:252/channel:1629282842.212, detail: [Aug 18 10:34:02] DEBUG[13746] stasis.c: Topic 'cache:252/channel:1629282842.212': 0x7f0c8002f820 created [Aug 18 10:34:02] DEBUG[13746] channel.c: Channel 0x7f0c80065e60 'Snoop/213007-0000000d' allocated [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13836] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13790] channel.c: Channel 0x7f0c8408c060 'Announcer/ARI-00000025;1' allocated [Aug 18 10:34:02] DEBUG[13790] stasis.c: Creating topic. name: channel:1629282842.213, detail: [Aug 18 10:34:02] DEBUG[13790] stasis.c: Topic 'channel:1629282842.213': 0x7f0c8407dfc0 created [Aug 18 10:34:02] DEBUG[13790] stasis.c: Creating topic. name: cache:253/channel:1629282842.213, detail: [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Finding handler for 213065 [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking channels create: Didn't match 213065 [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13832] res_ari.c: Checking channels externalMedia: Didn't match 213065 [Aug 18 10:34:02] DEBUG[13832] res_ari.c: No explicit handler found for 213065. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13703] channel.c: Channel 0x7f0c24016f20 'Announcer/ARI-0000001f;2' destroying [Aug 18 10:34:02] DEBUG[13703] stasis.c: Destroying topic. name: cache:219/channel:1629282839.184, detail: [Aug 18 10:34:02] DEBUG[13703] stasis.c: Topic 'cache:219/channel:1629282839.184': 0x7f0c240f8cc0 destroyed [Aug 18 10:34:02] DEBUG[13509] channel.c: Channel 0x7f0c2005c6f0 'Announcer/ARI-00000016;1' destroying [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:02] DEBUG[13836] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13746] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13790] stasis.c: Topic 'cache:253/channel:1629282842.213': 0x7f0c840682a0 created [Aug 18 10:34:02] DEBUG[13746] http.c: HTTP closing session. Top level [Aug 18 10:34:02] DEBUG[13837] http.c: HTTP Request URI is /ari/channels/213069?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116971&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13750] channel.c: Channel 0x7f0c9400a450 'Recorder/ARI-00000024;2' allocated [Aug 18 10:34:02] DEBUG[13837] http.c: match request [ari/channels/213069] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13666] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d'. Checking compatability for channels 'SIP/zvonobot-0000002a' and 'Recorder/ARI-00000020;2' [Aug 18 10:34:02] DEBUG[13666] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as channel 'SIP/zvonobot-0000002a' has features which prevent it [Aug 18 10:34:02] DEBUG[13666] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:02] DEBUG[13666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13666] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:02] DEBUG[13666] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:34:02] DEBUG[13703] stasis.c: Destroying topic. name: channel:1629282839.184, detail: [Aug 18 10:34:02] DEBUG[13750] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13846] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13703] stasis.c: Topic 'channel:1629282839.184': 0x7f0c24049eb0 destroyed [Aug 18 10:34:02] DEBUG[13837] http.c: match request [ari/channels/213069] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13846] http.c: HTTP Request URI is /ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play?media=sound%3Asilence%2F2 [Aug 18 10:34:02] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13840] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13837] http.c: match request [ari/channels/213069] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Finding handler for channels/213068 [Aug 18 10:34:02] DEBUG[13852] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is joining [Aug 18 10:34:02] DEBUG[13846] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13851] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13825] http.c: match request [ari/channels/213067] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13839] stasis/app.c: Channel '1629282842.212' is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13840] http.c: HTTP Request URI is /ari/channels/213070?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116970&callerId=74950493843 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 From: ;tag=as6ceaa437 To: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag [Aug 18 10:34:02] DEBUG[13846] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Finding handler for 213068 [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking channels create: Didn't match 213068 [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13836] res_ari.c: Checking channels externalMedia: Didn't match 213068 [Aug 18 10:34:02] DEBUG[13836] res_ari.c: No explicit handler found for 213068. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 864, ms is 74 [Aug 18 10:34:02] DEBUG[13854] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13846] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:02] DEBUG[13854] http.c: HTTP Request URI is /ari/channels/213072?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116968&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13854] http.c: match request [ari/channels/213072] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13849] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13851] http.c: HTTP Request URI is /ari/channels/213071?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116969&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13837] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13840] http.c: match request [ari/channels/213070] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213007&app=calls_0&format=slin16&external_host=127.0.0.1%3A50353 [Aug 18 10:34:02] DEBUG[13825] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13846] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13509] stasis.c: Destroying topic. name: cache:163/channel:1629282835.136, detail: [Aug 18 10:34:02] DEBUG[13509] stasis.c: Topic 'cache:163/channel:1629282835.136': 0x7f0c2003b580 destroyed [Aug 18 10:34:02] DEBUG[13509] stasis.c: Destroying topic. name: channel:1629282835.136, detail: [Aug 18 10:34:02] DEBUG[13509] stasis.c: Topic 'channel:1629282835.136': 0x7f0c2003b430 destroyed [Aug 18 10:34:02] DEBUG[13840] http.c: match request [ari/channels/213070] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Finding handler for bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Finding handler for bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Finding handler for aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13846] res_ari.c: No explicit handler found for aba705f1-c39f-408a-8a02-8c7f66ee7c7d. Using wildcard bridgeId. [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Finding handler for play [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:02] DEBUG[13855] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13854] http.c: match request [ari/channels/213072] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP got report of 100 bytes from 178.62.121.41:11671 [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:02] DEBUG[13846] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:02] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] http.c: match request [ari/channels/213072] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13846] stasis.c: Creating topic. name: channel:1629282842.214, detail: [Aug 18 10:34:02] DEBUG[13846] stasis.c: Topic 'channel:1629282842.214': 0x7f0c280d1ac0 created [Aug 18 10:34:02] DEBUG[13846] stasis.c: Creating topic. name: cache:254/channel:1629282842.214, detail: [Aug 18 10:34:02] DEBUG[13846] stasis.c: Topic 'cache:254/channel:1629282842.214': 0x7f0c280da0f0 created [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13840] http.c: match request [ari/channels/213070] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13837] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13852] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pushing 0x7f0c940389d0(Recorder/ARI-00000024;2) [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: Allocating new SIP dialog for 7e7c331c1fe2791e5fd406d43558f841@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13832] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c123480' [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) RTP allocated port 19412 [Aug 18 10:34:02] DEBUG[13849] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13851] http.c: match request [ari/channels/213071] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: Allocating new SIP dialog for 37143b113692b4be24257efe4a622e06@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13818] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac01e130' [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) RTP allocated port 15986 [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE creating session 0.0.0.0:15986 (15986) [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE create [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add system candidates [Aug 18 10:34:02] DEBUG[13818] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13818] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 159.65.48.104:15986, 2130706431 [Aug 18 10:34:02] DEBUG[13818] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13818] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 10.131.0.10:15986, 2130706431 [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13855] http.c: HTTP Request URI is /ari/channels/213073?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116967&callerId=74950493843 [Aug 18 10:34:02] DEBUG[13854] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13840] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE creating session 0.0.0.0:19412 (19412) [Aug 18 10:34:02] DEBUG[13825] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE create [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE add system candidates [Aug 18 10:34:02] DEBUG[13832] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13832] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE add candidate: 159.65.48.104:19412, 2130706431 [Aug 18 10:34:02] DEBUG[13832] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13832] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE add candidate: 10.131.0.10:19412, 2130706431 [Aug 18 10:34:02] DEBUG[13832] rtp_engine.c: RTP instance '0x7f0c1c123480' is setup and ready to go [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) ICE stopped [Aug 18 10:34:02] DEBUG[13832] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13832] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13855] http.c: match request [ari/channels/213073] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13855] http.c: match request [ari/channels/213073] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13851] http.c: match request [ari/channels/213071] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516628 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821117000@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d Max-Forwards: 70 From: ;tag=as6093d024 To: Contact: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117000@178.62.121.41", nonce="40ce9240", response="5eb350c3459ef31d13146b9eacb9d70e" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 274160452 274160453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946687 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (1) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318273 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13818] rtp_engine.c: RTP instance '0x7f0cac01e130' is setup and ready to go [Aug 18 10:34:02] DEBUG[13840] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13854] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Finding handler for channels/213069 [Aug 18 10:34:02] DEBUG[13832] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: Allocating new SIP dialog for 25edde7d1b9da1cc0838ece2672ef65c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13822] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c100f6c90' [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) RTP allocated port 10716 [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE creating session 0.0.0.0:10716 (10716) [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE create [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE add system candidates [Aug 18 10:34:02] DEBUG[13822] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13822] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE add candidate: 159.65.48.104:10716, 2130706431 [Aug 18 10:34:02] DEBUG[13822] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13822] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE add candidate: 10.131.0.10:10716, 2130706431 [Aug 18 10:34:02] DEBUG[13822] rtp_engine.c: RTP instance '0x7f0c100f6c90' is setup and ready to go [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE stopped [Aug 18 10:34:02] DEBUG[13822] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13822] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13822] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13822] res_rtp_asterisk.c: (0x7f0c100f6c90) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13822] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13822] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: SIP call-id changed from '25edde7d1b9da1cc0838ece2672ef65c@127.0.1.1:5060' to '769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13822] stasis.c: Creating topic. name: channel:213064, detail: [Aug 18 10:34:02] DEBUG[13822] stasis.c: Topic 'channel:213064': 0x7f0c10041cc0 created [Aug 18 10:34:02] DEBUG[13822] stasis.c: Creating topic. name: cache:255/channel:213064, detail: [Aug 18 10:34:02] DEBUG[13822] stasis.c: Topic 'cache:255/channel:213064': 0x7f0c100723a0 created [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13851] http.c: match request [ari/channels/213071] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) ICE stopped [Aug 18 10:34:02] DEBUG[13818] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13818] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13818] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13818] res_rtp_asterisk.c: (0x7f0cac01e130) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13818] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13818] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13818] chan_sip.c: SIP call-id changed from '37143b113692b4be24257efe4a622e06@127.0.1.1:5060' to '2c5322d560d5755f39711b55002aec77@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13818] stasis.c: Creating topic. name: channel:213066, detail: [Aug 18 10:34:02] DEBUG[13818] stasis.c: Topic 'channel:213066': 0x7f0cac044a30 created [Aug 18 10:34:02] DEBUG[13818] stasis.c: Creating topic. name: cache:256/channel:213066, detail: [Aug 18 10:34:02] DEBUG[13818] stasis.c: Topic 'cache:256/channel:213066': 0x7f0cac0320a0 created [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] http.c: match request [ari/channels/213073] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Finding handler for channels/213070 [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Finding handler for 213070 [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking channels create: Didn't match 213070 [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13840] res_ari.c: Checking channels externalMedia: Didn't match 213070 [Aug 18 10:34:02] DEBUG[13840] res_ari.c: No explicit handler found for 213070. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Finding handler for channels/213072 [Aug 18 10:34:02] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13832] res_rtp_asterisk.c: (0x7f0c1c123480) RTCP setup on RTP instance [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13851] http.c: Match made with [ari] [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13851] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13852] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:02] VERBOSE[13852] bridge_channel.c: Channel Recorder/ARI-00000024;2 joined 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: Allocating new SIP dialog for 06cc60ff20d627db7078ba650da7f99d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] VERBOSE[13832] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13836] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c180cf000' [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Finding handler for 213069 [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) RTP allocated port 12928 [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE creating session 0.0.0.0:12928 (12928) [Aug 18 10:34:02] DEBUG[13792] channel.c: Channel 0x7f0c7807b3b0 'UnicastRTP/127.0.0.1:50430-0x7f0c7804a920' allocated [Aug 18 10:34:02] DEBUG[13792] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:02] VERBOSE[13792] res_rtp_asterisk.c: 0x7f0c780818f0 -- Strict RTP learning after remote address set to: 127.0.0.1:50430 [Aug 18 10:34:02] DEBUG[13792] res_stasis.c: calls_0: Subscribing to robot_213023 [Aug 18 10:34:02] DEBUG[13792] stasis/app.c: Channel 'robot_213023' is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:02] DEBUG[13356] channel.c: Channel 0x7f0c240501f0 'Recorder/ARI-0000000d;1' destroying [Aug 18 10:34:02] DEBUG[13753] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280' [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking channels create: Didn't match 213069 [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13832] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13852] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677'. Checking compatability for channels 'SIP/zvonobot-00000013' and 'Recorder/ARI-00000024;2' [Aug 18 10:34:02] DEBUG[13852] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as could not get details [Aug 18 10:34:02] DEBUG[13852] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:02] DEBUG[13852] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13852] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13852] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:02] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677 is already using the new technology. [Aug 18 10:34:02] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is joining simple_bridge technology [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel Recorder/ARI-00000024;2 setting read format path: slin -> slin [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel SIP/zvonobot-00000013 setting write format path: slin -> ulaw [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel SIP/zvonobot-00000013 setting read format path: ulaw -> slin [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: slin -> slin [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Finding handler for channels/213067 [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Finding handler for 213067 [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking channels create: Didn't match 213067 [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13825] res_ari.c: Checking channels externalMedia: Didn't match 213067 [Aug 18 10:34:02] DEBUG[13825] res_ari.c: No explicit handler found for 213067. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13753] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:02] DEBUG[13753] http.c: HTTP closing session. Top level [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13792] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:02] DEBUG[13356] stasis.c: Destroying topic. name: cache:118/channel:1629282833.99, detail: [Aug 18 10:34:02] DEBUG[13356] stasis.c: Topic 'cache:118/channel:1629282833.99': 0x7f0c24048530 destroyed [Aug 18 10:34:02] DEBUG[13356] stasis.c: Destroying topic. name: channel:1629282833.99, detail: [Aug 18 10:34:02] DEBUG[13356] stasis.c: Topic 'channel:1629282833.99': 0x7f0c24006850 destroyed [Aug 18 10:34:02] DEBUG[13855] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13360] channel.c: Channel 0x7f0c700195c0 'SIP/zvonobot-00000009' destroying [Aug 18 10:34:02] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 640, ms is 60 [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13837] res_ari.c: Checking channels externalMedia: Didn't match 213069 [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:02] DEBUG[20620] stasis/app.c: channel '212973': is 0 interested in calls_0 [Aug 18 10:34:02] DEBUG[20620] stasis/app.c: channel '212973' unsubscribed from calls_0 [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Finding handler for 213072 [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE create [Aug 18 10:34:02] DEBUG[13837] res_ari.c: No explicit handler found for 213069. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13666] audiohook.c: Audiohook 0x7f0c80063290 has stale audio in its factories. Flushing them both [Aug 18 10:34:02] DEBUG[13855] http.c: HTTP consuming request body [Aug 18 10:34:02] DEBUG[13792] http.c: HTTP closing session. Top level [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Finding handler for channels/213071 [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE add system candidates [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] VERBOSE[13857] dial.c: Called 127.0.0.1:50430 [Aug 18 10:34:02] VERBOSE[13857] dial.c: UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 answered [Aug 18 10:34:02] VERBOSE[13857] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 [Aug 18 10:34:02] DEBUG[13857] stasis/app.c: Channel 'robot_213023' is 2 interested in calls_0 [Aug 18 10:34:02] DEBUG[13858] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking channels create: Didn't match 213072 [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13858] http.c: HTTP Request URI is /ari/channels/212999 [Aug 18 10:34:02] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Finding handler for channels/213073 [Aug 18 10:34:02] DEBUG[13564] bridge_channel.c: Setting 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) state from:0 to:1 [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13854] res_ari.c: Checking channels externalMedia: Didn't match 213072 [Aug 18 10:34:02] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 688, ms is 63 [Aug 18 10:34:02] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282842.217, detail: [Aug 18 10:34:02] DEBUG[13360] stasis.c: Destroying topic. name: cache:16/channel:212973, detail: [Aug 18 10:34:02] DEBUG[13360] stasis.c: Topic 'cache:16/channel:212973': 0x7f0c7007f5c0 destroyed [Aug 18 10:34:02] DEBUG[13360] stasis.c: Destroying topic. name: channel:212973, detail: [Aug 18 10:34:02] DEBUG[13360] stasis.c: Topic 'channel:212973': 0x7f0c70080140 destroyed [Aug 18 10:34:02] DEBUG[13360] channel.c: Channel 0x7f0c40046650 'Snoop/212973-00000005' destroying [Aug 18 10:34:02] DEBUG[13360] stasis.c: Destroying topic. name: cache:122/channel:1629282833.101, detail: [Aug 18 10:34:02] DEBUG[13360] stasis.c: Topic 'cache:122/channel:1629282833.101': 0x7f0c40029460 destroyed [Aug 18 10:34:02] DEBUG[13360] stasis.c: Destroying topic. name: channel:1629282833.101, detail: [Aug 18 10:34:02] DEBUG[13360] stasis.c: Topic 'channel:1629282833.101': 0x7f0c40006720 destroyed [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'channel:1629282842.217': 0x7f0c3002e830 created [Aug 18 10:34:02] DEBUG[20545] stasis.c: Creating topic. name: cache:257/channel:1629282842.217, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'cache:257/channel:1629282842.217': 0x7f0c300700f0 created [Aug 18 10:34:02] DEBUG[20545] stasis.c: Destroying topic. name: cache:257/channel:1629282842.217, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'cache:257/channel:1629282842.217': 0x7f0c300700f0 destroyed [Aug 18 10:34:02] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282842.217, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'channel:1629282842.217': 0x7f0c3002e830 destroyed [Aug 18 10:34:02] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000009', '', 'Stasis', 'calls_0', 18, 7, 'ANSWERED', 3, '', '212973', '')] [Aug 18 10:34:02] DEBUG[13836] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13836] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE add candidate: 159.65.48.104:12928, 2130706431 [Aug 18 10:34:02] DEBUG[13836] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13836] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE add candidate: 10.131.0.10:12928, 2130706431 [Aug 18 10:34:02] DEBUG[13836] rtp_engine.c: RTP instance '0x7f0c180cf000' is setup and ready to go [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) ICE stopped [Aug 18 10:34:02] DEBUG[13836] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13836] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13836] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13836] res_rtp_asterisk.c: (0x7f0c180cf000) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13836] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13836] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282842.218, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'channel:1629282842.218': 0x7f0c3002e830 created [Aug 18 10:34:02] DEBUG[20545] stasis.c: Creating topic. name: cache:258/channel:1629282842.218, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'cache:258/channel:1629282842.218': 0x7f0c300112d0 created [Aug 18 10:34:02] DEBUG[20545] stasis.c: Destroying topic. name: cache:258/channel:1629282842.218, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'cache:258/channel:1629282842.218': 0x7f0c300112d0 destroyed [Aug 18 10:34:02] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282842.218, detail: [Aug 18 10:34:02] DEBUG[20545] stasis.c: Topic 'channel:1629282842.218': 0x7f0c3002e830 destroyed [Aug 18 10:34:02] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'Snoop/212973-00000005', 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660', 'Stasis', 'calls_0', 7, 7, 'ANSWERED', 3, '', '1629282833.101', '')] [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13564] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: pulling 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] http.c: match request [ari/channels/212999] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:02] VERBOSE[13564] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 left 'simple_bridge' stasis-bridge [Aug 18 10:34:02] DEBUG[13564] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) is leaving simple_bridge technology [Aug 18 10:34:02] DEBUG[13564] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' can not use native RTP bridge as two channels are required [Aug 18 10:34:02] DEBUG[13564] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:02] DEBUG[13564] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13564] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13564] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:02] DEBUG[13564] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 is already using the new technology. [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13854] res_ari.c: No explicit handler found for 213072. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212999-00000008 - start 1629282835.575127 answer 1629282835.575127 end 1629282842.477755 dur 6.902 bill 6.902 dispo ANSWERED [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Finding handler for 213073 [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] http.c: match request [ari/channels/212999] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking channels create: Didn't match 213073 [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13855] res_ari.c: Checking channels externalMedia: Didn't match 213073 [Aug 18 10:34:02] DEBUG[13855] res_ari.c: No explicit handler found for 213073. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13858] http.c: match request [ari/channels/212999] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13858] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13832] chan_sip.c: SIP call-id changed from '7e7c331c1fe2791e5fd406d43558f841@127.0.1.1:5060' to '25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:02] DEBUG[13836] chan_sip.c: SIP call-id changed from '06cc60ff20d627db7078ba650da7f99d@127.0.1.1:5060' to '43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Finding handler for channels/212999 [Aug 18 10:34:02] DEBUG[13836] stasis.c: Creating topic. name: channel:213068, detail: [Aug 18 10:34:02] DEBUG[13860] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13564] bridge_channel.c: Bridge is returning 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) to write format slin16 [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13812] stasis.c: Creating topic. name: channel:1629282842.221, detail: [Aug 18 10:34:02] DEBUG[13812] stasis.c: Topic 'channel:1629282842.221': 0x7f0cb00635c0 created [Aug 18 10:34:02] DEBUG[13812] stasis.c: Creating topic. name: cache:259/channel:1629282842.221, detail: [Aug 18 10:34:02] DEBUG[13812] stasis.c: Topic 'cache:259/channel:1629282842.221': 0x7f0cb00a6490 created [Aug 18 10:34:02] DEBUG[13564] channel.c: Channel UnicastRTP/127.0.0.1:50116-0x7f0c24077280 setting write format path: slin16 -> slin16 [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13832] stasis.c: Creating topic. name: channel:213065, detail: [Aug 18 10:34:02] DEBUG[13832] stasis.c: Topic 'channel:213065': 0x7f0c1c0575d0 created [Aug 18 10:34:02] DEBUG[13832] stasis.c: Creating topic. name: cache:260/channel:213065, detail: [Aug 18 10:34:02] DEBUG[13832] stasis.c: Topic 'cache:260/channel:213065': 0x7f0c1c0b08f0 created [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13849] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:02] DEBUG[13849] netsock2.c: Splitting '127.0.0.1:50353' into... [Aug 18 10:34:02] DEBUG[13849] netsock2.c: ...host '127.0.0.1' and port '50353'. [Aug 18 10:34:02] DEBUG[13849] netsock2.c: Splitting '127.0.0.1:50353' into... [Aug 18 10:34:02] DEBUG[13849] netsock2.c: ...host '127.0.0.1' and port '50353'. [Aug 18 10:34:02] DEBUG[13849] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13836] stasis.c: Topic 'channel:213068': 0x7f0c180a9610 created [Aug 18 10:34:02] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP audio difference is 776, ms is 117 [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706391 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13564] stasis/control.c: robot_212999, c66c6480-4085-4bd9-87d2-ee6f5748dcc3: Channel was departed from bridge [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13564] stasis/app.c: bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3': is 3 interested in calls_0 [Aug 18 10:34:02] DEBUG[13860] http.c: HTTP Request URI is /ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc/addChannel?channel=1629282840.199%2Crobot_213023 [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13508] stasis/control.c: robot_212999: Channel departing bridge [Aug 18 10:34:02] DEBUG[13508] bridge.c: Waiting for 0x7f0c840887e0(UnicastRTP/127.0.0.1:50116-0x7f0c24077280) bridge thread to die. [Aug 18 10:34:02] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 744, ms is 113 [Aug 18 10:34:02] DEBUG[13750] res_stasis_recording.c: 1629282840.195: Sending record(212982_SjNdOhkmKEHHUVlGZbmvYPyClcnqXyTA.wav) command [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: Allocating new SIP dialog for 3024cf406e39c2191c38c32158a874e1@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13837] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c24122cb0' [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) RTP allocated port 13558 [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE creating session 0.0.0.0:13558 (13558) [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE create [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE add system candidates [Aug 18 10:34:02] DEBUG[13837] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13837] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE add candidate: 159.65.48.104:13558, 2130706431 [Aug 18 10:34:02] DEBUG[13837] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13837] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE add candidate: 10.131.0.10:13558, 2130706431 [Aug 18 10:34:02] DEBUG[13837] rtp_engine.c: RTP instance '0x7f0c24122cb0' is setup and ready to go [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE stopped [Aug 18 10:34:02] DEBUG[13837] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13837] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13836] stasis.c: Creating topic. name: cache:261/channel:213068, detail: [Aug 18 10:34:02] DEBUG[13849] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c340f6d00' [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP allocated port 12792 [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE creating session 127.0.0.1:12792 (12792) [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE create [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE add system candidates [Aug 18 10:34:02] DEBUG[13849] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13849] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE add candidate: 159.65.48.104:12792, 2130706431 [Aug 18 10:34:02] DEBUG[13849] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13849] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13849] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE add candidate: 10.131.0.10:12792, 2130706431 [Aug 18 10:34:02] DEBUG[13849] rtp_engine.c: RTP instance '0x7f0c340f6d00' is setup and ready to go [Aug 18 10:34:02] DEBUG[13849] stasis.c: Creating topic. name: channel:robot_213007, detail: [Aug 18 10:34:02] DEBUG[13849] stasis.c: Topic 'channel:robot_213007': 0x7f0c340fce20 created [Aug 18 10:34:02] DEBUG[13849] stasis.c: Creating topic. name: cache:262/channel:robot_213007, detail: [Aug 18 10:34:02] DEBUG[13849] stasis.c: Topic 'cache:262/channel:robot_213007': 0x7f0c3402ad50 created [Aug 18 10:34:02] DEBUG[13863] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13750] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13750] http.c: HTTP closing session. Top level [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 832, ms is 72 [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Finding handler for 213071 [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 768, ms is 68 [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking channels create: Didn't match 213071 [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Finding handler for 212999 [Aug 18 10:34:02] DEBUG[13860] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13860] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13860] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc/addChannel] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13860] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Finding handler for bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc/addChannel [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Finding handler for bridges [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking channels create: Didn't match 212999 [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: Allocating new SIP dialog for 648b4efb57297c225366d6243d809e09@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13855] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c4005ac00' [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) RTP allocated port 18778 [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE creating session 0.0.0.0:18778 (18778) [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE create [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE add system candidates [Aug 18 10:34:02] DEBUG[13855] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13855] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE add candidate: 159.65.48.104:18778, 2130706431 [Aug 18 10:34:02] DEBUG[13855] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13855] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE add candidate: 10.131.0.10:18778, 2130706431 [Aug 18 10:34:02] DEBUG[13855] rtp_engine.c: RTP instance '0x7f0c4005ac00' is setup and ready to go [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE stopped [Aug 18 10:34:02] DEBUG[13855] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13855] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13855] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13855] res_rtp_asterisk.c: (0x7f0c4005ac00) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13855] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13855] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13855] chan_sip.c: SIP call-id changed from '648b4efb57297c225366d6243d809e09@127.0.1.1:5060' to '39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13855] stasis.c: Creating topic. name: channel:213073, detail: [Aug 18 10:34:02] DEBUG[13855] stasis.c: Topic 'channel:213073': 0x7f0c40048e60 created [Aug 18 10:34:02] DEBUG[13855] stasis.c: Creating topic. name: cache:263/channel:213073, detail: [Aug 18 10:34:02] DEBUG[13855] stasis.c: Topic 'cache:263/channel:213073': 0x7f0c400498e0 created [Aug 18 10:34:02] DEBUG[13771] stasis.c: Creating topic. name: channel:1629282842.223, detail: [Aug 18 10:34:02] DEBUG[13771] stasis.c: Topic 'channel:1629282842.223': 0x7f0c2c07ec50 created [Aug 18 10:34:02] DEBUG[13771] stasis.c: Creating topic. name: cache:264/channel:1629282842.223, detail: [Aug 18 10:34:02] DEBUG[13771] stasis.c: Topic 'cache:264/channel:1629282842.223': 0x7f0c2c0ac570 created [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel Recorder/ARI-00000024;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:02] DEBUG[13837] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:02] DEBUG[13851] res_ari.c: Checking channels externalMedia: Didn't match 213071 [Aug 18 10:34:02] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 1152, ms is 92 [Aug 18 10:34:02] DEBUG[13564] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13863] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:02] DEBUG[13837] res_rtp_asterisk.c: (0x7f0c24122cb0) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13837] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13837] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13837] chan_sip.c: SIP call-id changed from '3024cf406e39c2191c38c32158a874e1@127.0.1.1:5060' to '3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13837] stasis.c: Creating topic. name: channel:213069, detail: [Aug 18 10:34:02] DEBUG[13837] stasis.c: Topic 'channel:213069': 0x7f0c240f94f0 created [Aug 18 10:34:02] DEBUG[13837] stasis.c: Creating topic. name: cache:265/channel:213069, detail: [Aug 18 10:34:02] DEBUG[13837] stasis.c: Topic 'cache:265/channel:213069': 0x7f0c240f9f70 created [Aug 18 10:34:02] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: Allocating new SIP dialog for 594138803d6d257476a6614766494f34@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13840] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c0a9e10' [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) RTP allocated port 14444 [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE creating session 0.0.0.0:14444 (14444) [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE create [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add system candidates [Aug 18 10:34:02] DEBUG[13840] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13840] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add candidate: 159.65.48.104:14444, 2130706431 [Aug 18 10:34:02] DEBUG[13840] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13840] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add candidate: 10.131.0.10:14444, 2130706431 [Aug 18 10:34:02] DEBUG[13840] rtp_engine.c: RTP instance '0x7f0c2c0a9e10' is setup and ready to go [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE stopped [Aug 18 10:34:02] DEBUG[13840] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13840] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13840] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13840] res_rtp_asterisk.c: (0x7f0c2c0a9e10) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13840] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13840] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13840] chan_sip.c: SIP call-id changed from '594138803d6d257476a6614766494f34@127.0.1.1:5060' to '31084e6149d402b41e86a7dd14209045@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13840] stasis.c: Creating topic. name: channel:213070, detail: [Aug 18 10:34:02] DEBUG[13840] stasis.c: Topic 'channel:213070': 0x7f0c2c0cad10 created [Aug 18 10:34:02] DEBUG[13840] stasis.c: Creating topic. name: cache:266/channel:213070, detail: [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:02] DEBUG[13851] res_ari.c: No explicit handler found for 213071. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13840] stasis.c: Topic 'cache:266/channel:213070': 0x7f0c2c0cb790 created [Aug 18 10:34:02] DEBUG[13858] res_ari.c: Checking channels externalMedia: Didn't match 212999 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: Allocating new SIP dialog for 51e3716217231dbe20f46c08331a340b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13825] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c0802d370' [Aug 18 10:34:02] DEBUG[13508] stasis/app.c: channel 'robot_212999': is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13852] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: alaw -> slin [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #14 (4) INVITE - 5 [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13508] channel.c: Channel 0x7f0c240f0910 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280' hanging up. Refs: 2 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:02] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #14)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #70 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #70)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (2) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668828 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:02] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13858] res_ari.c: No explicit handler found for 212999. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13785] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 832, ms is 72 [Aug 18 10:34:02] DEBUG[13863] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP audio difference is 776, ms is 117 [Aug 18 10:34:02] DEBUG[13836] stasis.c: Topic 'cache:261/channel:213068': 0x7f0c180c9110 created [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:02] DEBUG[13780] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660' [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) RTP allocated port 11962 [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE creating session 0.0.0.0:11962 (11962) [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE create [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE add system candidates [Aug 18 10:34:02] DEBUG[13825] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13825] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13385] bridge_channel.c: Setting 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) state from:0 to:1 [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13846] channel.c: Channel 0x7f0c280a4fb0 'Announcer/ARI-00000026;1' allocated [Aug 18 10:34:02] DEBUG[13846] stasis.c: Creating topic. name: channel:1629282842.227, detail: [Aug 18 10:34:02] DEBUG[13846] stasis.c: Topic 'channel:1629282842.227': 0x7f0c280f4310 created [Aug 18 10:34:02] DEBUG[13846] stasis.c: Creating topic. name: cache:267/channel:1629282842.227, detail: [Aug 18 10:34:02] DEBUG[13846] stasis.c: Topic 'cache:267/channel:1629282842.227': 0x7f0c280ec8f0 created [Aug 18 10:34:02] DEBUG[13385] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: pulling 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) [Aug 18 10:34:02] DEBUG[13863] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13780] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:02] DEBUG[13780] http.c: HTTP closing session. Top level [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13864] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13864] http.c: HTTP Request URI is /ari/channels/212973 [Aug 18 10:34:02] VERBOSE[13385] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50139-0x7f0c8c042660 left 'simple_bridge' stasis-bridge <6f6fd705-00c9-4b9f-a75f-d7c933b85e66> [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13861] app.c: play_and_record: , /var/spool/asterisk/recording/212982_SjNdOhkmKEHHUVlGZbmvYPyClcnqXyTA, 'wav' [Aug 18 10:34:02] DEBUG[13861] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:02] VERBOSE[13861] app.c: x=0, open writing: /var/spool/asterisk/recording/212982_SjNdOhkmKEHHUVlGZbmvYPyClcnqXyTA format: wav, 0x7f0c78075720 [Aug 18 10:34:02] DEBUG[13864] http.c: match request [ari/channels/212973] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13385] bridge_channel.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) is leaving simple_bridge technology [Aug 18 10:34:02] DEBUG[13790] channel.c: Channel 0x7f0c8410b600 'Announcer/ARI-00000025;2' allocated [Aug 18 10:34:02] DEBUG[13790] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:02] DEBUG[13790] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000025;1' [Aug 18 10:34:02] DEBUG[13866] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c8408a2a0(Announcer/ARI-00000025;2) is joining [Aug 18 10:34:02] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 816, ms is 71 [Aug 18 10:34:02] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 672, ms is 62 [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE add candidate: 159.65.48.104:11962, 2130706431 [Aug 18 10:34:02] DEBUG[13825] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13785] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as0a75a671 Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" Content-Length: 0 <-------------> [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a75a671 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as0a75a671 [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13385] bridge_native_rtp.c: Bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' can not use native RTP bridge as two channels are required [Aug 18 10:34:02] DEBUG[13385] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:02] DEBUG[13385] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13863] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13863] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Finding handler for bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Finding handler for bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:02] DEBUG[13863] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:02] DEBUG[13863] stasis.c: Creating topic. name: bridge:a76fe935-dd52-4012-a523-638ab1ec4dfe, detail: [Aug 18 10:34:02] DEBUG[13863] stasis.c: Topic 'bridge:a76fe935-dd52-4012-a523-638ab1ec4dfe': 0x7f0c8408b9f0 created [Aug 18 10:34:02] DEBUG[13863] stasis.c: Creating topic. name: cache:268/bridge:a76fe935-dd52-4012-a523-638ab1ec4dfe, detail: [Aug 18 10:34:02] DEBUG[13863] stasis.c: Topic 'cache:268/bridge:a76fe935-dd52-4012-a523-638ab1ec4dfe': 0x7f0c8408d0a0 created [Aug 18 10:34:02] DEBUG[13864] http.c: match request [ari/channels/212973] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13385] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:02] DEBUG[13385] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:02] DEBUG[13385] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 is already using the new technology. [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:02] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 864, ms is 74 [Aug 18 10:34:02] DEBUG[13863] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' can not use native RTP bridge as two channels are required [Aug 18 10:34:02] DEBUG[13863] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:02] DEBUG[13863] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:02] DEBUG[13863] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:02] DEBUG[13863] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:02] DEBUG[13863] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: calling simple_bridge technology constructor [Aug 18 10:34:02] DEBUG[13863] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: calling simple_bridge technology start [Aug 18 10:34:02] DEBUG[13863] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:02] DEBUG[13864] http.c: match request [ari/channels/212973] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13864] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Finding handler for channels/212973 [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Finding handler for 212973 [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking channels create: Didn't match 212973 [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13864] res_ari.c: Checking channels externalMedia: Didn't match 212973 [Aug 18 10:34:02] DEBUG[13864] res_ari.c: No explicit handler found for 212973. Using wildcard channelId. [Aug 18 10:34:02] DEBUG[13863] http.c: HTTP closing session. Top level [Aug 18 10:34:02] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 1168, ms is 166 [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Finding handler for 382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Stopping retransmission on '16541045053beef31ed8e5361337aa22@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:02] DEBUG[13825] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE add candidate: 10.131.0.10:11962, 2130706431 [Aug 18 10:34:02] DEBUG[13825] rtp_engine.c: RTP instance '0x7f0c0802d370' is setup and ready to go [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) ICE stopped [Aug 18 10:34:02] DEBUG[13825] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13825] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13825] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13867] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:02] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:02] DEBUG[13860] res_ari.c: No explicit handler found for 382ca601-8f64-4a7e-bdde-fe8fb07c61bc. Using wildcard bridgeId. [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Finding handler for addChannel [Aug 18 10:34:02] DEBUG[13860] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13860] stasis/control.c: 1629282840.199: Sending channel add_to_bridge command [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac Max-Forwards: 70 From: ;tag=as79336d5f To: ;tag=as0a75a671 Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:02] DEBUG[13785] bridge_roles.c: Roles did not exist on channel Snoop/213023-0000000c [Aug 18 10:34:02] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 672, ms is 62 [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13825] res_rtp_asterisk.c: (0x7f0c0802d370) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13825] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:02] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13385] stasis/control.c: robot_212973, 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: Channel was departed from bridge [Aug 18 10:34:02] DEBUG[13867] http.c: HTTP Request URI is /ari/channels/212982/snoop?app=calls_0&spy=in [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13385] stasis/app.c: bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66': is 2 interested in calls_0 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:02] DEBUG[13375] stasis/control.c: robot_212973: Channel departing bridge [Aug 18 10:34:02] DEBUG[13375] bridge.c: Waiting for 0x7f0c9403d460(UnicastRTP/127.0.0.1:50139-0x7f0c8c042660) bridge thread to die. [Aug 18 10:34:02] DEBUG[13785] stasis/control.c: 1629282840.199: Adding to bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:02] DEBUG[13785] stasis/app.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:02] DEBUG[13385] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:02] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13375] stasis/app.c: channel 'robot_212973': is 1 interested in calls_0 [Aug 18 10:34:02] DEBUG[13375] channel.c: Channel 0x7f0c8c057560 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660' hanging up. Refs: 2 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:02] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[13825] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13825] chan_sip.c: SIP call-id changed from '51e3716217231dbe20f46c08331a340b@127.0.1.1:5060' to '70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060' [Aug 18 10:34:02] DEBUG[13825] stasis.c: Creating topic. name: channel:213067, detail: [Aug 18 10:34:02] DEBUG[13825] stasis.c: Topic 'channel:213067': 0x7f0c0806bd50 created [Aug 18 10:34:02] DEBUG[13825] stasis.c: Creating topic. name: cache:269/channel:213067, detail: [Aug 18 10:34:02] DEBUG[13825] stasis.c: Topic 'cache:269/channel:213067': 0x7f0c0803d410 created [Aug 18 10:34:02] DEBUG[13866] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pushing 0x7f0c8408a2a0(Announcer/ARI-00000025;2) [Aug 18 10:34:02] DEBUG[13869] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: 0x7f0c7c0842d0(Snoop/213023-0000000c) is joining [Aug 18 10:34:02] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13867] http.c: match request [ari/channels/212982/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13870] http.c: HTTP opening session. Top level [Aug 18 10:34:02] DEBUG[13857] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Audio is at 15904 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682952 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13870] http.c: HTTP Request URI is /ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13870] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [httpstatus] len 10 [Aug 18 10:34:02] DEBUG[13867] http.c: match request [ari/channels/212982/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[13867] http.c: match request [ari/channels/212982/snoop] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13870] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [phoneprov] len 9 [Aug 18 10:34:02] DEBUG[13870] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play] with handler [ari] len 3 [Aug 18 10:34:02] DEBUG[13867] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[13870] http.c: Match made with [ari] [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:02] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Finding handler for bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060/play [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116998@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69 Max-Forwards: 70 From: ;tag=as52d5bd88 To: Contact: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116998@178.62.121.41", nonce="6d1cff09", response="ee2e4f50590d744bdb64bb0ed2ff3ae6" Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 194010121 194010122 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11140 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Finding handler for bridges [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:02] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Finding handler for channels/212982/snoop [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Finding handler for channels [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:02] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13866] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:02] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:02] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #64 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[13869] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: pushing 0x7f0c7c0842d0(Snoop/213023-0000000c) [Aug 18 10:34:02] DEBUG[13822] channel.c: Channel 0x7f0c10040760 'SIP/zvonobot-00000064' allocated [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:02] DEBUG[13822] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #64)) [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:02] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:02] VERBOSE[13866] bridge_channel.c: Channel Announcer/ARI-00000025;2 joined 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407135 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (5) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:33:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771033 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (4) INVITE - 5 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Finding handler for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:34:02] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:02] VERBOSE[13869] bridge_channel.c: Channel Snoop/213023-0000000c joined 'simple_bridge' stasis-bridge <382ca601-8f64-4a7e-bdde-fe8fb07c61bc> [Aug 18 10:34:02] DEBUG[13867] res_ari.c: Finding handler for 212982 [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:02] DEBUG[13851] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:02] DEBUG[13851] chan_sip.c: Allocating new SIP dialog for 2815af0f0d97666f2285ab4457c5b410@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:02] DEBUG[13851] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c30021550' [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) RTP allocated port 15924 [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE creating session 0.0.0.0:15924 (15924) [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE create [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE add system candidates [Aug 18 10:34:02] DEBUG[13851] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:02] DEBUG[13851] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE add candidate: 159.65.48.104:15924, 2130706431 [Aug 18 10:34:02] DEBUG[13851] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:02] DEBUG[13851] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE add candidate: 10.131.0.10:15924, 2130706431 [Aug 18 10:34:02] DEBUG[13851] rtp_engine.c: RTP instance '0x7f0c30021550' is setup and ready to go [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) ICE stopped [Aug 18 10:34:02] DEBUG[13851] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:02] DEBUG[13851] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:02] DEBUG[13851] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:02] DEBUG[13851] res_rtp_asterisk.c: (0x7f0c30021550) RTCP setup on RTP instance [Aug 18 10:34:02] VERBOSE[13851] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:02] DEBUG[13851] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:02] DEBUG[13851] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:02] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (3) INVITE - 5 [Aug 18 10:34:02] DEBUG[13870] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[13870] res_ari.c: No explicit handler found for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channels create: Didn't match 212982 [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Finding handler for play [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13851] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:03] DEBUG[13866] bridge.c: Chose bridge technology softmix [Aug 18 10:34:03] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: switching from simple_bridge technology to softmix [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology constructor [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13851] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0ca803dbf0(SIP/zvonobot-0000003b) to dummy bridge temporarily [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0c94055bb0(Recorder/ARI-0000001e;2) to dummy bridge temporarily [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channels externalMedia: Didn't match 212982 [Aug 18 10:34:03] DEBUG[13867] res_ari.c: No explicit handler found for 212982. Using wildcard channelId. [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Finding handler for snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:03] DEBUG[13867] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:03] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:03] DEBUG[13818] channel.c: Channel 0x7f0cac032dc0 'SIP/zvonobot-00000065' allocated [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is leaving simple_bridge technology (dummy) [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:03] DEBUG[13812] channel.c: Channel 0x7f0cb010fb20 'Snoop/213009-0000000e' allocated [Aug 18 10:34:03] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:34:03] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:34:03] DEBUG[13851] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:03] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:03] DEBUG[13869] bridge_native_rtp.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology stop [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:03] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:34:03] DEBUG[13812] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13812] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13695] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3'. Checking compatability for channels 'SIP/zvonobot-0000002f' and 'Recorder/ARI-00000023;2' [Aug 18 10:34:03] DEBUG[13695] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' can not use native RTP bridge as channel 'SIP/zvonobot-0000002f' has features which prevent it [Aug 18 10:34:03] DEBUG[13695] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13695] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13695] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13695] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13695] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 is already using the new technology. [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c8408a2a0(Announcer/ARI-00000025;2) is joining softmix technology [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[13851] chan_sip.c: SIP call-id changed from '2815af0f0d97666f2285ab4457c5b410@127.0.1.1:5060' to '2485aced650f4f671041baca16773141@159.65.48.104:5060' [Aug 18 10:34:03] DEBUG[13851] stasis.c: Creating topic. name: channel:213071, detail: [Aug 18 10:34:03] DEBUG[13851] stasis.c: Topic 'channel:213071': 0x7f0c3010e130 created [Aug 18 10:34:03] DEBUG[13851] stasis.c: Creating topic. name: cache:270/channel:213071, detail: [Aug 18 10:34:03] DEBUG[13851] stasis.c: Topic 'cache:270/channel:213071': 0x7f0c3010eb40 created [Aug 18 10:34:03] DEBUG[13879] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13818] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Announcer/ARI-00000025;2: [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: Allocating new SIP dialog for 3c23afdb161610e36d56a60f517a7fd1@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:03] DEBUG[13854] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38082a80' [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) RTP allocated port 11106 [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE creating session 0.0.0.0:11106 (11106) [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE create [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE add system candidates [Aug 18 10:34:03] DEBUG[13854] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:03] DEBUG[13854] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE add candidate: 159.65.48.104:11106, 2130706431 [Aug 18 10:34:03] DEBUG[13854] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:03] DEBUG[13854] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE add candidate: 10.131.0.10:11106, 2130706431 [Aug 18 10:34:03] DEBUG[13854] rtp_engine.c: RTP instance '0x7f0c38082a80' is setup and ready to go [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) ICE stopped [Aug 18 10:34:03] DEBUG[13854] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:03] DEBUG[13854] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:03] DEBUG[13854] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:03] DEBUG[13854] res_rtp_asterisk.c: (0x7f0c38082a80) RTCP setup on RTP instance [Aug 18 10:34:03] VERBOSE[13854] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:03] DEBUG[13854] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:03] DEBUG[13854] chan_sip.c: SIP call-id changed from '3c23afdb161610e36d56a60f517a7fd1@127.0.1.1:5060' to '67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060' [Aug 18 10:34:03] DEBUG[13854] stasis.c: Creating topic. name: channel:213072, detail: [Aug 18 10:34:03] DEBUG[13854] stasis.c: Topic 'channel:213072': 0x7f0c3809b150 created [Aug 18 10:34:03] DEBUG[13854] stasis.c: Creating topic. name: cache:271/channel:213072, detail: [Aug 18 10:34:03] DEBUG[13854] stasis.c: Topic 'cache:271/channel:213072': 0x7f0c3809bbd0 created [Aug 18 10:34:03] DEBUG[13869] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 680, ms is 105 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:03] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50409 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13866] channel.c: Channel Announcer/ARI-00000025;2 setting write format path: slin -> slin [Aug 18 10:34:03] DEBUG[13869] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13879] http.c: HTTP Request URI is /ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play?media=sound%3Asilence%2F2 [Aug 18 10:34:03] DEBUG[13879] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13869] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:03] DEBUG[13884] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:03] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13869] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13879] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13869] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc is already using the new technology. [Aug 18 10:34:03] DEBUG[13882] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13818] res_stasis.c: calls_0: Subscribing to 213066 [Aug 18 10:34:03] DEBUG[13818] stasis/app.c: Channel '213066' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13879] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13822] res_stasis.c: calls_0: Subscribing to 213064 [Aug 18 10:34:03] DEBUG[13822] stasis/app.c: Channel '213064' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13879] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13869] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: 0x7f0c7c0842d0(Snoop/213023-0000000c) is joining simple_bridge technology [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Announcer/ARI-00000025;2: [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Outgoing Call for 79821116976 [Aug 18 10:34:03] DEBUG[13822] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:03] DEBUG[13818] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13818] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Finding handler for bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play [Aug 18 10:34:03] DEBUG[13884] http.c: HTTP Request URI is /ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:03] DEBUG[13882] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213009&app=calls_0&format=slin16&external_host=127.0.0.1%3A50264 [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Announcer/ARI-00000025;2: Not in SFU mode [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is joining softmix technology [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: SIP/zvonobot-0000003b: [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: SIP/zvonobot-0000003b: [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: SIP/zvonobot-0000003b: Not in SFU mode [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is joining softmix technology [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Recorder/ARI-0000001e;2: [Aug 18 10:34:03] DEBUG[13866] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: slin -> slin [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:03] DEBUG[13866] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Recorder/ARI-0000001e;2: [Aug 18 10:34:03] DEBUG[13866] bridge_softmix.c: Recorder/ARI-0000001e;2: Not in SFU mode [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology start [Aug 18 10:34:03] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology destructor [Aug 18 10:34:03] DEBUG[13870] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:03] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13822] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Outgoing Call for 79821116974 [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13860] stasis/control.c: robot_213023: Sending channel add_to_bridge command [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] DEBUG[13785] stasis/app.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' is 2 interested in calls_0 [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Finding handler for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13879] res_ari.c: No explicit handler found for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Finding handler for play [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:03] DEBUG[13879] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:03] DEBUG[13879] stasis.c: Creating topic. name: channel:1629282843.231, detail: [Aug 18 10:34:03] DEBUG[13879] stasis.c: Topic 'channel:1629282843.231': 0x7f0c980450b0 created [Aug 18 10:34:03] DEBUG[13879] stasis.c: Creating topic. name: cache:272/channel:1629282843.231, detail: [Aug 18 10:34:03] DEBUG[13879] stasis.c: Topic 'cache:272/channel:1629282843.231': 0x7f0c98045ae0 created [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] DEBUG[13884] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13884] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13873] stasis/app.c: Channel '1629282842.221' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13884] http.c: match request [ari/bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13695] audiohook.c: Audiohook 0x7f0cb011ab60 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13873] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 29 instead [Aug 18 10:34:03] DEBUG[13884] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13541] res_rtp_asterisk.c: (0x7f0c080871b0) RTP audio difference is 736, ms is 66 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:03] DEBUG[13882] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 From: ;tag=as6093d024 To: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Finding handler for bridges/d0f9af3e-7f00-4d11-8990-3d67ba7213d6/play [Aug 18 10:34:03] DEBUG[13882] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #21 - INVITE (got response) [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 From: ;tag=as63ca65c0 To: ;tag=as56bda81b Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56bda81b [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag as56bda81b [Aug 18 10:34:03] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 17 instead [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13882] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #44 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '63c7430f0c6f8039194559375e330124@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1 Max-Forwards: 70 From: ;tag=as63ca65c0 To: ;tag=as56bda81b Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Audio is at 14624 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:03] DEBUG[12869] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP 0x7f0cb400c820 -- Received packet from 178.62.121.41:14926, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[12869] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP 0x7f0cb400c820 -- Received packet from 178.62.121.41:14926, dropping due to strict RTP protection. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13882] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Finding handler for d0f9af3e-7f00-4d11-8990-3d67ba7213d6 [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13884] res_ari.c: No explicit handler found for d0f9af3e-7f00-4d11-8990-3d67ba7213d6. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Finding handler for play [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:03] DEBUG[13884] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[13885] chan_sip.c: Audio is at 15986 [Aug 18 10:34:03] VERBOSE[13885] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[13885] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[13885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Initializing initreq for method INVITE - callid 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116974@178.62.121.41 SIP/2.0 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 3 [ 52]: From: ;tag=as5b87d923 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 6 [ 60]: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:03 GMT [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] VERBOSE[13885] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:34:03] DEBUG[13885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 From: ;tag=as1dc5ead1 To: ;tag=as741846ee Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1dc5ead1 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as741846ee [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Finding handler for channels [Aug 18 10:34:03] DEBUG[13832] channel.c: Channel 0x7f0c1c12fa80 'SIP/zvonobot-00000066' allocated [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:03] DEBUG[13832] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:03] DEBUG[13263] res_rtp_asterisk.c: (0x7f0c24032c40) RTP 0x7f0c2403c460 -- Received packet from 178.62.121.41:10694, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13263] res_rtp_asterisk.c: (0x7f0c24032c40) RTP 0x7f0c2403c460 -- Received packet from 178.62.121.41:10694, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:03] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] DEBUG[13790] res_stasis_playback.c: 1629282841.210: Sending play(sound:silence/2) command [Aug 18 10:34:03] DEBUG[13888] channel.c: Channel Announcer/ARI-00000025;1 setting write format path: gsm -> slin [Aug 18 10:34:03] DEBUG[13888] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:03] VERBOSE[13888] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:03] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 (Checking To) --From tag as1dc5ead1 --To-tag as741846ee [Aug 18 10:34:03] DEBUG[13790] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:03] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] VERBOSE[13885] dial.c: Called zvonobot/79821116974 [Aug 18 10:34:03] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:03] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13790] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13872] bridge_softmix.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: starting mixing thread [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:03] VERBOSE[13887] chan_sip.c: Audio is at 10716 [Aug 18 10:34:03] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP ooh, format changed from none to ulaw [Aug 18 10:34:03] VERBOSE[13627] res_rtp_asterisk.c: 0x7f0c7c037ee0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:10224 [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13882] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13832] res_stasis.c: calls_0: Subscribing to 213065 [Aug 18 10:34:03] VERBOSE[13887] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] DEBUG[13832] stasis/app.c: Channel '213065' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Outgoing Call for 79821116975 [Aug 18 10:34:03] DEBUG[13832] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13882] netsock2.c: Splitting '127.0.0.1:50264' into... [Aug 18 10:34:03] DEBUG[13873] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 640, ms is 60 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP got report of 100 bytes from 178.62.121.41:10225 [Aug 18 10:34:03] DEBUG[13832] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:03] DEBUG[12869] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP 0x7f0cb400c820 -- Received packet from 178.62.121.41:14926, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[12869] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP 0x7f0cb400c820 -- Received packet from 178.62.121.41:14926, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[12869] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP 0x7f0cb400c820 -- Received packet from 178.62.121.41:14926, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] VERBOSE[13887] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] DEBUG[13882] netsock2.c: ...host '127.0.0.1' and port '50264'. [Aug 18 10:34:03] DEBUG[13882] netsock2.c: Splitting '127.0.0.1:50264' into... [Aug 18 10:34:03] DEBUG[13882] netsock2.c: ...host '127.0.0.1' and port '50264'. [Aug 18 10:34:03] DEBUG[13882] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:03] DEBUG[13882] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca003db80' [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) RTP allocated port 16202 [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) ICE creating session 127.0.0.1:16202 (16202) [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) ICE create [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) ICE add system candidates [Aug 18 10:34:03] DEBUG[13882] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:03] DEBUG[13882] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) ICE add candidate: 159.65.48.104:16202, 2130706431 [Aug 18 10:34:03] DEBUG[13882] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:03] DEBUG[13882] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:03] DEBUG[13882] res_rtp_asterisk.c: (0x7f0ca003db80) ICE add candidate: 10.131.0.10:16202, 2130706431 [Aug 18 10:34:03] DEBUG[13882] rtp_engine.c: RTP instance '0x7f0ca003db80' is setup and ready to go [Aug 18 10:34:03] DEBUG[13882] stasis.c: Creating topic. name: channel:robot_213009, detail: [Aug 18 10:34:03] DEBUG[13882] stasis.c: Topic 'channel:robot_213009': 0x7f0ca00df2d0 created [Aug 18 10:34:03] DEBUG[13882] stasis.c: Creating topic. name: cache:273/channel:robot_213009, detail: [Aug 18 10:34:03] DEBUG[13882] stasis.c: Topic 'cache:273/channel:robot_213009': 0x7f0ca00dfda0 created [Aug 18 10:34:03] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13857] stasis/control.c: robot_213023: Adding to bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:03] DEBUG[13857] stasis/app.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' is 3 interested in calls_0 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] VERBOSE[13887] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Initializing initreq for method INVITE - callid 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116976@178.62.121.41 SIP/2.0 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 3 [ 52]: From: ;tag=as7a3cd0ea [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 6 [ 60]: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:03 GMT [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] VERBOSE[13887] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #80 [Aug 18 10:34:03] DEBUG[13887] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: ;tag=as741846ee Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:03] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] VERBOSE[13890] chan_sip.c: Audio is at 19412 [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] VERBOSE[13890] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13891] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: 0x7f0c74093920(UnicastRTP/127.0.0.1:50430-0x7f0c7804a920) is joining [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[13890] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] DEBUG[13263] res_rtp_asterisk.c: (0x7f0c24032c40) RTP 0x7f0c2403c460 -- Received packet from 178.62.121.41:10694, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Audio is at 10612 [Aug 18 10:34:03] VERBOSE[13890] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[13887] dial.c: Called zvonobot/79821116976 [Aug 18 10:34:03] DEBUG[13263] res_rtp_asterisk.c: (0x7f0c24032c40) RTP 0x7f0c2403c460 -- Received packet from 178.62.121.41:10694, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13263] res_rtp_asterisk.c: (0x7f0c24032c40) RTP 0x7f0c2403c460 -- Received packet from 178.62.121.41:10694, dropping due to strict RTP protection. [Aug 18 10:34:03] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Initializing initreq for method INVITE - callid 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116975@178.62.121.41 SIP/2.0 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 3 [ 52]: From: ;tag=as5a5dd50f [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:03] DEBUG[13459] audiohook.c: Audiohook 0x7f0cac057320 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 6 [ 60]: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13891] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: pushing 0x7f0c74093920(UnicastRTP/127.0.0.1:50430-0x7f0c7804a920) [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:03 GMT [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #40 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #70 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #70)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] VERBOSE[13890] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Aug 18 10:34:03] DEBUG[13890] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (1) INVITE - 5 [Aug 18 10:34:03] VERBOSE[13890] dial.c: Called zvonobot/79821116975 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 From: ;tag=as02885f54 To: ;tag=as5580464d Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as02885f54 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5580464d [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 (Checking To) --From tag as02885f54 --To-tag as5580464d [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '10507dcf059680b46ad884550335c862@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:03] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0 Max-Forwards: 70 From: ;tag=as02885f54 To: ;tag=as5580464d Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Audio is at 14750 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #51 (5) BYE - 8 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #51)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: BYE sip:79821117054@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d Max-Forwards: 70 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117054@178.62.121.41:5060", nonce="171898dd", response="42493e75abf4f76e54b48a1bc7ad580d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[13891] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 joined 'simple_bridge' stasis-bridge <382ca601-8f64-4a7e-bdde-fe8fb07c61bc> [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 From: ;tag=as0261f463 To: ;tag=as64b58d1a Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 460639390 460639390 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0261f463 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64b58d1a [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 460639390 460639390 IN IP4 178.62.121.41 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17848 RTP/AVP 0 8 101 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking To) --From tag as0261f463 --To-tag as64b58d1a [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Got SDP version 460639390 and unique parts [root 460639390 IN IP4 178.62.121.41] [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 460639390 460639390 IN IP4 178.62.121.41... OK. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:03] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:03] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:03] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:03] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) ICE set role failed; no ice instance [Aug 18 10:34:03] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13849] channel.c: Channel 0x7f0c34028b90 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00' allocated [Aug 18 10:34:03] DEBUG[13849] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:03] VERBOSE[13849] res_rtp_asterisk.c: 0x7f0c340e36f0 -- Strict RTP learning after remote address set to: 127.0.0.1:50353 [Aug 18 10:34:03] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 640, ms is 60 [Aug 18 10:34:03] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13771] channel.c: Channel 0x7f0c2c0b7210 'Snoop/213011-0000000f' allocated [Aug 18 10:34:03] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:03] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 - start 1629282842.408403 answer 1629282842.439583 end 1629282843.390908 dur 0.982 bill 0.951 dispo ANSWERED [Aug 18 10:34:03] DEBUG[13891] bridge_native_rtp.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc'. Checking compatability for channels 'Snoop/213023-0000000c' and 'UnicastRTP/127.0.0.1:50430-0x7f0c7804a920' [Aug 18 10:34:03] DEBUG[13891] bridge_native_rtp.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' can not use native RTP bridge as could not get details [Aug 18 10:34:03] DEBUG[13891] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13891] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13891] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13891] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13771] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13771] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13849] res_stasis.c: calls_0: Subscribing to robot_213007 [Aug 18 10:34:03] DEBUG[13849] stasis/app.c: Channel 'robot_213007' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13849] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13849] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13900] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13900] http.c: HTTP Request URI is /ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play?media=sound%3Asilence%2F2 [Aug 18 10:34:03] DEBUG[13900] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13900] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 9 instead [Aug 18 10:34:03] DEBUG[13839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) RTCP setting address on RTP instance [Aug 18 10:34:03] DEBUG[13891] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc is already using the new technology. [Aug 18 10:34:03] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c88023770 -- Strict RTP learning after remote address set to: 178.62.121.41:17848 [Aug 18 10:34:03] DEBUG[13619] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52'. Checking compatability for channels 'SIP/zvonobot-00000030' and 'Recorder/ARI-0000001c;2' [Aug 18 10:34:03] DEBUG[13619] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as channel 'SIP/zvonobot-00000030' has features which prevent it [Aug 18 10:34:03] DEBUG[13619] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13619] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13619] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13619] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13619] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 is already using the new technology. [Aug 18 10:34:03] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:34:03] DEBUG[13906] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:17848 [Aug 18 10:34:03] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00da918) from 0x7f0c147e2330 to 0x7f0c88003ff8 [Aug 18 10:34:03] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00a6508) from 0x7f0c147e2330 to 0x7f0c88003ff8 [Aug 18 10:34:03] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb01084f8) from 0x7f0c147e2330 to 0x7f0c88003ff8 [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) RTCP ignoring duplicate property [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:03] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001c setting read format path: alaw -> alaw [Aug 18 10:34:03] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001c setting write format path: alaw -> alaw [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS - ast_rtp_activate rtp=0x7f0c88023770 - setup and perform DTLS' [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88023770) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88023770) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:03] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:03] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Strict routing enforced for session 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:03] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:03] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117049@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3ff04840 Max-Forwards: 70 From: ;tag=as0261f463 To: ;tag=as64b58d1a Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13900] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [ari] len 3 [Aug 18 10:34:03] VERBOSE[13047] dial.c: SIP/zvonobot-0000001c answered [Aug 18 10:34:03] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 672, ms is 62 [Aug 18 10:34:03] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] VERBOSE[13047] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000001c [Aug 18 10:34:03] DEBUG[13906] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213011&app=calls_0&format=slin16&external_host=127.0.0.1%3A50349 [Aug 18 10:34:03] DEBUG[13047] stasis/app.c: Channel '212991' is 2 interested in calls_0 [Aug 18 10:34:03] VERBOSE[13047] res_rtp_asterisk.c: 0x7f0c88023770 -- Strict RTP switching to RTP target address 178.62.121.41:17848 as source [Aug 18 10:34:03] DEBUG[13047] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:03] DEBUG[13047] channel.c: Channel SIP/zvonobot-0000001c setting read format path: ulaw -> alaw [Aug 18 10:34:03] DEBUG[13047] channel.c: Channel SIP/zvonobot-0000001c setting write format path: alaw -> ulaw [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:03] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13891] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: 0x7f0c74093920(UnicastRTP/127.0.0.1:50430-0x7f0c7804a920) is joining simple_bridge technology [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 setting read format path: slin16 -> slin16 [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel Snoop/213023-0000000c setting write format path: slin16 -> slin [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel Snoop/213023-0000000c setting read format path: slin -> slin16 [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 setting write format path: slin16 -> slin16 [Aug 18 10:34:03] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13837] channel.c: Channel 0x7f0c24110ce0 'SIP/zvonobot-00000069' allocated [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:03] DEBUG[13837] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:03] DEBUG[13508] res_rtp_asterisk.c: (0x7f0c24077280) DTLS stop [Aug 18 10:34:03] DEBUG[13508] res_rtp_asterisk.c: (0x7f0c24077280) DTLS srtp - stopped timeout timer' [Aug 18 10:34:03] DEBUG[13508] res_rtp_asterisk.c: (0x7f0c24077280) ICE RTP transport deallocating [Aug 18 10:34:03] DEBUG[13508] res_rtp_asterisk.c: (0x7f0c24077280) ICE stopped [Aug 18 10:34:03] DEBUG[13508] rtp_engine.c: Destroyed RTP instance '0x7f0c24077280' [Aug 18 10:34:03] DEBUG[13508] channel.c: Channel 0x7f0c240f0910 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280' destroying [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:03] DEBUG[20620] stasis/app.c: channel 'robot_212999': is 0 interested in calls_0 [Aug 18 10:34:03] DEBUG[20620] stasis/app.c: channel 'robot_212999' unsubscribed from calls_0 [Aug 18 10:34:03] DEBUG[13900] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Session timer started: 42 - 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 1768000ms [Aug 18 10:34:03] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282843.233, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'channel:1629282843.233': 0x7f0c300b3650 created [Aug 18 10:34:03] DEBUG[20545] stasis.c: Creating topic. name: cache:274/channel:1629282843.233, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'cache:274/channel:1629282843.233': 0x7f0c3007f570 created [Aug 18 10:34:03] DEBUG[13508] stasis.c: Destroying topic. name: cache:170/channel:robot_212999, detail: [Aug 18 10:34:03] DEBUG[13508] stasis.c: Topic 'cache:170/channel:robot_212999': 0x7f0c240f27e0 destroyed [Aug 18 10:34:03] DEBUG[13508] stasis.c: Destroying topic. name: channel:robot_212999, detail: [Aug 18 10:34:03] DEBUG[13508] stasis.c: Topic 'channel:robot_212999': 0x7f0c240f2580 destroyed [Aug 18 10:34:03] DEBUG[20545] stasis.c: Destroying topic. name: cache:274/channel:1629282843.233, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'cache:274/channel:1629282843.233': 0x7f0c3007f570 destroyed [Aug 18 10:34:03] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282843.233, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'channel:1629282843.233': 0x7f0c300b3650 destroyed [Aug 18 10:34:03] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:56', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280', '', 'Stasis', 'calls_0', 1, 1, 'ANSWERED', 3, '', 'robot_212999', '')] [Aug 18 10:34:03] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Finding handler for bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Finding handler for e594e1d1-53fe-4904-8517-472d8e3b8b52 [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13900] res_ari.c: No explicit handler found for e594e1d1-53fe-4904-8517-472d8e3b8b52. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Finding handler for play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:03] DEBUG[13900] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:03] DEBUG[13900] stasis.c: Creating topic. name: channel:1629282843.234, detail: [Aug 18 10:34:03] DEBUG[13900] stasis.c: Topic 'channel:1629282843.234': 0x7f0c1c136420 created [Aug 18 10:34:03] DEBUG[13900] stasis.c: Creating topic. name: cache:275/channel:1629282843.234, detail: [Aug 18 10:34:03] DEBUG[13900] stasis.c: Topic 'cache:275/channel:1629282843.234': 0x7f0c1c136e80 created [Aug 18 10:34:03] DEBUG[13906] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13906] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13837] res_stasis.c: calls_0: Subscribing to 213069 [Aug 18 10:34:03] DEBUG[13837] stasis/app.c: Channel '213069' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13837] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP got report of 100 bytes from 178.62.121.41:12913 [Aug 18 10:34:03] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1200, ms is 95 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Outgoing Call for 79821116971 [Aug 18 10:34:03] DEBUG[13906] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13837] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 1056, ms is 86 [Aug 18 10:34:03] DEBUG[13906] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP creating BEGIN DTMF Frame: 53 (5), at 178.62.121.41:11670 [Aug 18 10:34:03] DTMF[13056] channel.c: DTMF begin '5' received on SIP/zvonobot-00000000 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:03] VERBOSE[13914] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 From: ;tag=as2b432b30 To: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e9aea7f;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2b432b30 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13860] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:03] DEBUG[13860] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:03] DTMF[13056] channel.c: DTMF begin passthrough '5' on SIP/zvonobot-00000000 [Aug 18 10:34:03] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13896] stasis/app.c: Channel '1629282842.223' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:03] DEBUG[13857] stasis/app.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' is 4 interested in calls_0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] DTMF[13153] channel.c: DTMF begin '5' received on Announcer/ARI-00000002;1 [Aug 18 10:34:03] DTMF[13153] channel.c: DTMF begin ignored '5' on Announcer/ARI-00000002;1 [Aug 18 10:34:03] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 1048, ms is 151 [Aug 18 10:34:03] DTMF[13059] channel.c: DTMF begin '5' received on Recorder/ARI-00000000;1 [Aug 18 10:34:03] DTMF[13059] channel.c: DTMF begin passthrough '5' on Recorder/ARI-00000000;1 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] VERBOSE[13905] dial.c: Called 127.0.0.1:50353 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060 (Checking To) --From tag as2b432b30 --To-tag [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:03] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 1288, ms is 181 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '799e753c40b511cf0e89de4f0a586534@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:03] DEBUG[13891] channel.c: Channel UnicastRTP/127.0.0.1:50430-0x7f0c7804a920 setting write format path: slin -> slin16 [Aug 18 10:34:03] DEBUG[13896] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 46 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP ooh, format changed from none to slin16 [Aug 18 10:34:03] VERBOSE[13905] dial.c: UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 answered [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] VERBOSE[13905] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 [Aug 18 10:34:03] DEBUG[13905] stasis/app.c: Channel 'robot_213007' is 2 interested in calls_0 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:03] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13933] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13933] http.c: HTTP Request URI is /ari/bridges/beb17a84-adfc-4fa3-b7a8-31977a540c1f/addChannel?channel=1629282842.212%2Crobot_213007 [Aug 18 10:34:03] DEBUG[13933] http.c: match request [ari/bridges/beb17a84-adfc-4fa3-b7a8-31977a540c1f/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13933] http.c: match request [ari/bridges/beb17a84-adfc-4fa3-b7a8-31977a540c1f/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13933] http.c: match request [ari/bridges/beb17a84-adfc-4fa3-b7a8-31977a540c1f/addChannel] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13933] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Finding handler for bridges/beb17a84-adfc-4fa3-b7a8-31977a540c1f/addChannel [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13770] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #31 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #31)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Finding handler for channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13906] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:03] DEBUG[13906] netsock2.c: Splitting '127.0.0.1:50349' into... [Aug 18 10:34:03] DEBUG[13906] netsock2.c: ...host '127.0.0.1' and port '50349'. [Aug 18 10:34:03] DEBUG[13906] netsock2.c: Splitting '127.0.0.1:50349' into... [Aug 18 10:34:03] DEBUG[13906] netsock2.c: ...host '127.0.0.1' and port '50349'. [Aug 18 10:34:03] DEBUG[13906] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:03] DEBUG[13906] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c240f8a30' [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP allocated port 15964 [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE creating session 127.0.0.1:15964 (15964) [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE create [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE add system candidates [Aug 18 10:34:03] DEBUG[13906] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:03] DEBUG[13906] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE add candidate: 159.65.48.104:15964, 2130706431 [Aug 18 10:34:03] DEBUG[13906] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:03] DEBUG[13906] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:03] DEBUG[13906] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE add candidate: 10.131.0.10:15964, 2130706431 [Aug 18 10:34:03] DEBUG[13906] rtp_engine.c: RTP instance '0x7f0c240f8a30' is setup and ready to go [Aug 18 10:34:03] DEBUG[13906] stasis.c: Creating topic. name: channel:robot_213011, detail: [Aug 18 10:34:03] DEBUG[13906] stasis.c: Topic 'channel:robot_213011': 0x7f0c24075fe0 created [Aug 18 10:34:03] DEBUG[13906] stasis.c: Creating topic. name: cache:276/channel:robot_213011, detail: [Aug 18 10:34:03] DEBUG[13906] stasis.c: Topic 'cache:276/channel:robot_213011': 0x7f0c24075da0 created [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 From: ;tag=as3edf3f1c To: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP creating END DTMF Frame: 53 (5), at 178.62.121.41:11670 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 494 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 494 [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DTMF[13056] channel.c: DTMF end '5' received on SIP/zvonobot-00000000, duration 160 ms [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:03] DTMF[13056] channel.c: DTMF end accepted with begin '5' on SIP/zvonobot-00000000 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] DEBUG[13928] http.c: HTTP opening session. Top level [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DTMF[13056] channel.c: DTMF end passthrough '5' on SIP/zvonobot-00000000 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[13920] chan_sip.c: Audio is at 13558 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:34:03] VERBOSE[13920] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] DEBUG[13928] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31c5c295 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] VERBOSE[13920] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] DTMF[13153] channel.c: DTMF end '5' received on Announcer/ARI-00000002;1, duration 160 ms [Aug 18 10:34:03] DTMF[13153] channel.c: DTMF end passthrough '5' on Announcer/ARI-00000002;1 [Aug 18 10:34:03] DTMF[13059] channel.c: DTMF end '5' received on Recorder/ARI-00000000;1, duration 160 ms [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag as31c5c295 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:03] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117048@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1 Max-Forwards: 70 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Contact: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:03] DTMF[13059] channel.c: DTMF end accepted with begin '5' on Recorder/ARI-00000000;1 [Aug 18 10:34:03] VERBOSE[13050] dial.c: SIP/zvonobot-0000001d is busy [Aug 18 10:34:03] DEBUG[13050] channel.c: Channel 0x7f0c9001fe80 'SIP/zvonobot-0000001d' hanging up. Refs: 2 [Aug 18 10:34:03] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001d - start 1629282826.244267 answer 0.000000 end 1629282843.692583 dur 17.448 bill 1629282843.692 dispo BUSY [Aug 18 10:34:03] DEBUG[13928] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:03] DTMF[13059] channel.c: DTMF end passthrough '5' on Recorder/ARI-00000000;1 [Aug 18 10:34:03] VERBOSE[13920] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (4) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Finding handler for beb17a84-adfc-4fa3-b7a8-31977a540c1f [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13933] res_ari.c: No explicit handler found for beb17a84-adfc-4fa3-b7a8-31977a540c1f. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Finding handler for addChannel [Aug 18 10:34:03] DEBUG[13933] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:03] DEBUG[13933] stasis/control.c: 1629282842.212: Sending channel add_to_bridge command [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13928] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #31 (2) INVITE - 5 [Aug 18 10:34:03] DEBUG[13928] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13928] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #31)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Initializing initreq for method INVITE - callid 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13839] bridge_roles.c: Roles did not exist on channel Snoop/213007-0000000d [Aug 18 10:34:03] DEBUG[13839] stasis/control.c: 1629282842.212: Adding to bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f [Aug 18 10:34:03] DEBUG[13839] stasis/app.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116971@178.62.121.41 SIP/2.0 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 3 [ 52]: From: ;tag=as611ff9f7 [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 6 [ 60]: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 From: ;tag=as6093d024 To: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13935] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c20083ee0(Snoop/213007-0000000d) is joining [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13928] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTCP got report of 100 bytes from 178.62.121.41:16139 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag [Aug 18 10:34:03] DEBUG[13928] stasis.c: Creating topic. name: bridge:5fd3583d-12a2-4028-9389-fce6801ffb6b, detail: [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:03 GMT [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13770] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13770] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13770] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13770] channel.c: Channel Announcer/ARI-00000021;1 setting write format path: slin -> slin [Aug 18 10:34:03] DEBUG[13770] channel.c: Channel 0x7f0c90063430 'Announcer/ARI-00000021;1' hanging up. Refs: 2 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[13928] stasis.c: Topic 'bridge:5fd3583d-12a2-4028-9389-fce6801ffb6b': 0x7f0c280d1970 created [Aug 18 10:34:03] DEBUG[13928] stasis.c: Creating topic. name: cache:277/bridge:5fd3583d-12a2-4028-9389-fce6801ffb6b, detail: [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] DEBUG[13778] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13928] stasis.c: Topic 'cache:277/bridge:5fd3583d-12a2-4028-9389-fce6801ffb6b': 0x7f0c280062a0 created [Aug 18 10:34:03] DEBUG[13778] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13778] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13778] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:03] DEBUG[13778] channel.c: Channel Announcer/ARI-00000022;1 setting write format path: slin -> slin [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:03] VERBOSE[13920] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #50 [Aug 18 10:34:03] DEBUG[13920] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13935] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: pushing 0x7f0c20083ee0(Snoop/213007-0000000d) [Aug 18 10:34:03] VERBOSE[13920] dial.c: Called zvonobot/79821116971 [Aug 18 10:34:03] DEBUG[13858] channel.c: Soft-Hanging (0x20) up channel 'SIP/zvonobot-00000023' [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 From: ;tag=as6ceaa437 To: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag [Aug 18 10:34:03] DEBUG[13858] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:03] DEBUG[13858] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Setting 0x7f0ca0053060(SIP/zvonobot-00000023) state from:0 to:1 [Aug 18 10:34:03] VERBOSE[13935] bridge_channel.c: Channel Snoop/213007-0000000d joined 'simple_bridge' stasis-bridge [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pulling 0x7f0ca0053060(SIP/zvonobot-00000023) [Aug 18 10:34:03] VERBOSE[13454] bridge_channel.c: Channel SIP/zvonobot-00000023 left 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0ca0053060(SIP/zvonobot-00000023) is leaving simple_bridge technology [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Setting 0x7f0c7c018d60(Recorder/ARI-00000013;2) state from:0 to:2 [Aug 18 10:34:03] DEBUG[13454] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] DEBUG[13454] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13454] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13454] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13454] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13454] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 is already using the new technology. [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Bridge is returning 0x7f0ca0053060(SIP/zvonobot-00000023) to read format alaw [Aug 18 10:34:03] DEBUG[13454] channel.c: Channel SIP/zvonobot-00000023 setting read format path: ulaw -> alaw [Aug 18 10:34:03] DEBUG[13454] bridge_channel.c: Bridge is returning 0x7f0ca0053060(SIP/zvonobot-00000023) to write format alaw [Aug 18 10:34:03] DEBUG[13454] channel.c: Channel SIP/zvonobot-00000023 setting write format path: alaw -> ulaw [Aug 18 10:34:03] DEBUG[13462] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: pulling 0x7f0c7c018d60(Recorder/ARI-00000013;2) [Aug 18 10:34:03] VERBOSE[13462] bridge_channel.c: Channel Recorder/ARI-00000013;2 left 'simple_bridge' stasis-bridge <3d1f9573-48d1-48a0-bcdd-9d7f09555162> [Aug 18 10:34:03] DEBUG[13462] bridge_channel.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: 0x7f0c7c018d60(Recorder/ARI-00000013;2) is leaving simple_bridge technology [Aug 18 10:34:03] DEBUG[13462] bridge_native_rtp.c: Bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] DEBUG[13462] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13462] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13462] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13462] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13462] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162 is already using the new technology. [Aug 18 10:34:03] DEBUG[13928] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000023 - start 1629282828.139390 answer 1629282835.159263 end 1629282843.782666 dur 15.643 bill 8.623 dispo ANSWERED [Aug 18 10:34:03] DEBUG[13928] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13454] stasis/control.c: 212999, 3d1f9573-48d1-48a0-bcdd-9d7f09555162: Channel was departed from bridge [Aug 18 10:34:03] DEBUG[13462] channel.c: Channel 0x7f0c7c017b90 'Recorder/ARI-00000013;2' hanging up. Refs: 2 [Aug 18 10:34:03] DEBUG[13454] stasis/app.c: bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162': is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13129] stasis/control.c: 212999: Channel departing bridge [Aug 18 10:34:03] DEBUG[13129] bridge.c: Waiting for 0x7f0ca0053060(SIP/zvonobot-00000023) bridge thread to die. [Aug 18 10:34:03] DEBUG[13778] channel.c: Channel 0x7f0c08044260 'Announcer/ARI-00000022;1' hanging up. Refs: 2 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[13454] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:03] DEBUG[13928] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13129] stasis/app.c: channel '212999': is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13129] channel.c: Channel 0x7f0c98012a90 'SIP/zvonobot-00000023' hanging up. Refs: 3 [Aug 18 10:34:03] DEBUG[13905] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 [Aug 18 10:34:03] DEBUG[13936] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13928] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:03] DEBUG[13928] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13928] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: calling simple_bridge technology constructor [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as2aed188c Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2aed188c [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13928] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: calling simple_bridge technology start [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as2aed188c [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '596ef69a675656833620d8345b47f33c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13928] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13935] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] DEBUG[13928] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13938] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13935] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13938] http.c: HTTP Request URI is /ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/addChannel?channel=212991 [Aug 18 10:34:03] DEBUG[13935] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13935] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13935] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13935] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f is already using the new technology. [Aug 18 10:34:03] DEBUG[13935] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c20083ee0(Snoop/213007-0000000d) is joining simple_bridge technology [Aug 18 10:34:03] DEBUG[13938] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 From: ;tag=as3f810040 To: ;tag=as622102a0 Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f810040 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as622102a0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13839] stasis/app.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' is 2 interested in calls_0 [Aug 18 10:34:03] DEBUG[13938] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13836] channel.c: Channel 0x7f0c180bb1b0 'SIP/zvonobot-00000067' allocated [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:03] DEBUG[13836] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:03] DEBUG[13933] stasis/control.c: robot_213007: Sending channel add_to_bridge command [Aug 18 10:34:03] DEBUG[13938] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/addChannel] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13938] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13936] http.c: HTTP Request URI is /ari/channels/1629282835.132 [Aug 18 10:34:03] DEBUG[13836] res_stasis.c: calls_0: Subscribing to 213068 [Aug 18 10:34:03] DEBUG[13836] stasis/app.c: Channel '213068' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13836] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] DEBUG[13836] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13936] http.c: match request [ari/channels/1629282835.132] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Finding handler for bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/addChannel [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Outgoing Call for 79821116972 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[13939] chan_sip.c: Audio is at 12928 [Aug 18 10:34:03] VERBOSE[13939] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[13939] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[13939] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Initializing initreq for method INVITE - callid 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116972@178.62.121.41 SIP/2.0 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 3 [ 52]: From: ;tag=as1ed67fff [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 6 [ 60]: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:03 GMT [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:03] VERBOSE[13939] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13936] http.c: match request [ari/channels/1629282835.132] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13936] http.c: match request [ari/channels/1629282835.132] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13936] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Finding handler for channels/1629282835.132 [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Finding handler for channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Finding handler for 1629282835.132 [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking channels create: Didn't match 1629282835.132 [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13936] res_ari.c: Checking channels externalMedia: Didn't match 1629282835.132 [Aug 18 10:34:03] DEBUG[13936] res_ari.c: No explicit handler found for 1629282835.132. Using wildcard channelId. [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Finding handler for 5fd3583d-12a2-4028-9389-fce6801ffb6b [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 (Checking To) --From tag as3f810040 --To-tag as622102a0 [Aug 18 10:34:03] DEBUG[13938] res_ari.c: No explicit handler found for 5fd3583d-12a2-4028-9389-fce6801ffb6b. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Finding handler for addChannel [Aug 18 10:34:03] DEBUG[13938] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:03] DEBUG[13938] stasis/control.c: 212991: Sending channel add_to_bridge command [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #55 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336 Max-Forwards: 70 From: ;tag=as3f810040 To: ;tag=as622102a0 Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Audio is at 15836 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #85 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (1) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 From: ;tag=as6093d024 To: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #83 [Aug 18 10:34:03] DEBUG[13939] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 From: ;tag=as34e313e7 To: ;tag=as51d2bd1a Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as34e313e7 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as51d2bd1a [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 (Checking To) --From tag as34e313e7 --To-tag as51d2bd1a [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (3) INVITE - 5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 From: ;tag=as2618d3a5 To: ;tag=as6e1d4afe Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" Content-Length: 0 <-------------> [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2618d3a5 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e1d4afe [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:03] DEBUG[13047] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000001c [Aug 18 10:34:03] DEBUG[13047] stasis/control.c: 212991: Adding to bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b [Aug 18 10:34:03] DEBUG[13047] stasis/app.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' is 1 interested in calls_0 [Aug 18 10:34:03] VERBOSE[13939] dial.c: Called zvonobot/79821116972 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:03] DEBUG[13941] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c9403d460(SIP/zvonobot-0000001c) is joining [Aug 18 10:34:03] DEBUG[13941] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pushing 0x7f0c9403d460(SIP/zvonobot-0000001c) [Aug 18 10:34:03] VERBOSE[13941] bridge_channel.c: Channel SIP/zvonobot-0000001c joined 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:03] DEBUG[13941] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:03] DEBUG[13941] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:03] DEBUG[13941] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13941] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13941] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:03] DEBUG[13941] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:03] DEBUG[13941] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c9403d460(SIP/zvonobot-0000001c) is joining simple_bridge technology [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 (Checking To) --From tag as2618d3a5 --To-tag as6e1d4afe [Aug 18 10:34:03] DEBUG[13864] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:03] DEBUG[13375] res_rtp_asterisk.c: (0x7f0c8c042660) DTLS stop [Aug 18 10:34:03] DEBUG[13375] res_rtp_asterisk.c: (0x7f0c8c042660) DTLS srtp - stopped timeout timer' [Aug 18 10:34:03] DEBUG[13375] res_rtp_asterisk.c: (0x7f0c8c042660) ICE RTP transport deallocating [Aug 18 10:34:03] DEBUG[13375] res_rtp_asterisk.c: (0x7f0c8c042660) ICE stopped [Aug 18 10:34:03] DEBUG[13375] rtp_engine.c: Destroyed RTP instance '0x7f0c8c042660' [Aug 18 10:34:03] DEBUG[13375] channel.c: Channel 0x7f0c8c057560 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660' destroying [Aug 18 10:34:03] DEBUG[13864] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282843.236, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'channel:1629282843.236': 0x7f0c300b3650 created [Aug 18 10:34:03] DEBUG[20545] stasis.c: Creating topic. name: cache:278/channel:1629282843.236, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'cache:278/channel:1629282843.236': 0x7f0c3011b3f0 created [Aug 18 10:34:03] DEBUG[13846] channel.c: Channel 0x7f0c280acfd0 'Announcer/ARI-00000026;2' allocated [Aug 18 10:34:03] DEBUG[13846] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:03] DEBUG[13846] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000026;1' [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:03] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:03] DEBUG[20620] stasis/app.c: channel 'robot_212973': is 0 interested in calls_0 [Aug 18 10:34:03] DEBUG[20620] stasis/app.c: channel 'robot_212973' unsubscribed from calls_0 [Aug 18 10:34:03] DEBUG[20620] stasis.c: Destroying topic. name: cache:127/channel:robot_212973, detail: [Aug 18 10:34:03] DEBUG[20620] stasis.c: Topic 'cache:127/channel:robot_212973': 0x7f0c8c059550 destroyed [Aug 18 10:34:03] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212973, detail: [Aug 18 10:34:03] DEBUG[20620] stasis.c: Topic 'channel:robot_212973': 0x7f0c8c059340 destroyed [Aug 18 10:34:03] DEBUG[20545] stasis.c: Destroying topic. name: cache:278/channel:1629282843.236, detail: [Aug 18 10:34:03] DEBUG[13944] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c280d1290(Announcer/ARI-00000026;2) is joining [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'cache:278/channel:1629282843.236': 0x7f0c3011b3f0 destroyed [Aug 18 10:34:03] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282843.236, detail: [Aug 18 10:34:03] DEBUG[20545] stasis.c: Topic 'channel:1629282843.236': 0x7f0c300b3650 destroyed [Aug 18 10:34:03] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50139-0x7f0c8c042660', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212973', '')] [Aug 18 10:34:03] DEBUG[13942] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13942] http.c: HTTP Request URI is /ari/channels/1629282833.101 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:03] DEBUG[13941] res_rtp_asterisk.c: (0x7f0c88003e20) RTP changing ssrc from 436681903 to 226810481 due to a source change [Aug 18 10:34:03] DEBUG[13047] stasis/app.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' is 2 interested in calls_0 [Aug 18 10:34:03] DEBUG[13938] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:03] DEBUG[13938] http.c: HTTP closing session. Top level [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #60 [Aug 18 10:34:03] DEBUG[13764] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13942] http.c: match request [ari/channels/1629282833.101] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Stopping retransmission on '68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf Max-Forwards: 70 From: ;tag=as2618d3a5 To: ;tag=as6e1d4afe Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Audio is at 17196 [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:03] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:03] DEBUG[13942] http.c: match request [ari/channels/1629282833.101] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13942] http.c: match request [ari/channels/1629282833.101] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13942] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Finding handler for channels/1629282833.101 [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Finding handler for channels [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #91 [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:03] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:03] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:03] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:03] DEBUG[13944] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pushing 0x7f0c280d1290(Announcer/ARI-00000026;2) [Aug 18 10:34:03] DEBUG[13944] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:03] VERBOSE[13944] bridge_channel.c: Channel Announcer/ARI-00000026;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:03] DEBUG[13945] http.c: HTTP opening session. Top level [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:03] DEBUG[13945] http.c: HTTP Request URI is /ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/record?name=212991_ThJucWjnLOCIgXIZwxMKjVMLdekVScRR&format=wav [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Finding handler for 1629282833.101 [Aug 18 10:34:03] DEBUG[13825] channel.c: Channel 0x7f0c080ee420 'SIP/zvonobot-0000006b' allocated [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:03] DEBUG[13825] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:03] DEBUG[13944] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking channels create: Didn't match 1629282833.101 [Aug 18 10:34:03] DEBUG[13944] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13944] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:03] DEBUG[13945] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/record] with handler [httpstatus] len 10 [Aug 18 10:34:03] DEBUG[13945] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/record] with handler [phoneprov] len 9 [Aug 18 10:34:03] DEBUG[13944] bridge.c: Chose bridge technology softmix [Aug 18 10:34:03] DEBUG[13945] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/record] with handler [ari] len 3 [Aug 18 10:34:03] DEBUG[13945] http.c: Match made with [ari] [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Finding handler for bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Finding handler for bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Finding handler for 5fd3583d-12a2-4028-9389-fce6801ffb6b [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:03] DEBUG[13945] res_ari.c: No explicit handler found for 5fd3583d-12a2-4028-9389-fce6801ffb6b. Using wildcard bridgeId. [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Finding handler for record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:03] DEBUG[13945] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:04] DEBUG[13905] stasis/control.c: robot_213007: Adding to bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f [Aug 18 10:34:04] DEBUG[13905] stasis/app.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' is 3 interested in calls_0 [Aug 18 10:34:04] DEBUG[13825] res_stasis.c: calls_0: Subscribing to 213067 [Aug 18 10:34:04] DEBUG[13825] stasis/app.c: Channel '213067' is 1 interested in calls_0 [Aug 18 10:34:03] DEBUG[13942] res_ari.c: Checking channels externalMedia: Didn't match 1629282833.101 [Aug 18 10:34:04] DEBUG[13942] res_ari.c: No explicit handler found for 1629282833.101. Using wildcard channelId. [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:04] DEBUG[13825] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:03] VERBOSE[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: switching from simple_bridge technology to softmix [Aug 18 10:34:04] DEBUG[13951] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13825] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13945] stasis.c: Creating topic. name: channel:1629282843.237, detail: [Aug 18 10:34:04] DEBUG[13851] channel.c: Channel 0x7f0c3010c400 'SIP/zvonobot-0000006c' allocated [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:04] DEBUG[13851] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:04] DEBUG[13851] res_stasis.c: calls_0: Subscribing to 213071 [Aug 18 10:34:04] DEBUG[13851] stasis/app.c: Channel '213071' is 1 interested in calls_0 [Aug 18 10:34:04] DEBUG[13851] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:04] DEBUG[13896] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[13851] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13951] http.c: HTTP Request URI is /ari/channels/213074?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116966&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Outgoing Call for 79821116973 [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Outgoing Call for 79821116969 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:04] VERBOSE[13950] chan_sip.c: Audio is at 11962 [Aug 18 10:34:04] VERBOSE[13950] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:04] VERBOSE[13950] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:04] VERBOSE[13950] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Initializing initreq for method INVITE - callid 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116973@178.62.121.41 SIP/2.0 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 3 [ 52]: From: ;tag=as6d27c109 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 6 [ 60]: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:04 GMT [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:04] VERBOSE[13950] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #92 [Aug 18 10:34:04] DEBUG[13950] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling softmix technology constructor [Aug 18 10:34:04] DEBUG[13951] http.c: match request [ari/channels/213074] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13947] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) is joining [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:04] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 688, ms is 63 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:04] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[13953] chan_sip.c: Audio is at 15924 [Aug 18 10:34:04] DEBUG[13951] http.c: match request [ari/channels/213074] with handler [phoneprov] len 9 [Aug 18 10:34:04] VERBOSE[13953] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:04] DEBUG[13945] stasis.c: Topic 'channel:1629282843.237': 0x7f0c7806efc0 created [Aug 18 10:34:04] DEBUG[13945] stasis.c: Creating topic. name: cache:279/channel:1629282843.237, detail: [Aug 18 10:34:04] DEBUG[13945] stasis.c: Topic 'cache:279/channel:1629282843.237': 0x7f0c78040590 created [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13951] http.c: match request [ari/channels/213074] with handler [ari] len 3 [Aug 18 10:34:04] VERBOSE[13953] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13951] http.c: Match made with [ari] [Aug 18 10:34:04] VERBOSE[13953] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:04] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP audio difference is 656, ms is 102 [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: moving 0x7f0c7006de00(SIP/zvonobot-0000002a) to dummy bridge temporarily [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Initializing initreq for method INVITE - callid 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116969@178.62.121.41 SIP/2.0 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 3 [ 52]: From: ;tag=as2f5156ef [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:04] VERBOSE[13950] dial.c: Called zvonobot/79821116973 [Aug 18 10:34:04] DEBUG[13951] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: moving 0x7f0c2c08b700(Recorder/ARI-00000020;2) to dummy bridge temporarily [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is leaving simple_bridge technology (dummy) [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:04] DEBUG[13957] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Finding handler for channels/213074 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 6 [ 60]: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology stop [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c280d1290(Announcer/ARI-00000026;2) is joining softmix technology [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Announcer/ARI-00000026;2: [Aug 18 10:34:04] DEBUG[13957] http.c: HTTP Request URI is /ari/channels/213076?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116964&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 800, ms is 70 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:04] DEBUG[13944] channel.c: Channel Announcer/ARI-00000026;2 setting write format path: slin -> slin [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13972] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13957] http.c: match request [ari/channels/213076] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:04] DEBUG[13541] res_rtp_asterisk.c: (0x7f0c080871b0) RTP audio difference is 928, ms is 78 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13967] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13957] http.c: match request [ari/channels/213076] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Finding handler for 213074 [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking channels create: Didn't match 213074 [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13951] res_ari.c: Checking channels externalMedia: Didn't match 213074 [Aug 18 10:34:04] DEBUG[13951] res_ari.c: No explicit handler found for 213074. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13972] http.c: HTTP Request URI is /ari/channels/213078?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116962&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13972] http.c: match request [ari/channels/213078] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:04] DEBUG[13961] http.c: HTTP Request URI is /ari/channels/213075?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116965&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 736, ms is 66 [Aug 18 10:34:04] DEBUG[13957] http.c: match request [ari/channels/213076] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13973] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:04 GMT [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:04] DEBUG[13967] http.c: HTTP Request URI is /ari/channels/213077?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116963&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: [Aug 18 10:34:04] DEBUG[13957] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13957] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Finding handler for channels/213076 [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Finding handler for 213076 [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking channels create: Didn't match 213076 [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13957] res_ari.c: Checking channels externalMedia: Didn't match 213076 [Aug 18 10:34:04] DEBUG[13957] res_ari.c: No explicit handler found for 213076. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13961] http.c: match request [ari/channels/213075] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:04] DEBUG[13972] http.c: match request [ari/channels/213078] with handler [phoneprov] len 9 [Aug 18 10:34:04] VERBOSE[13953] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #94 [Aug 18 10:34:04] DEBUG[13953] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:04] DEBUG[13967] http.c: match request [ari/channels/213077] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Announcer/ARI-00000026;2: [Aug 18 10:34:04] DEBUG[13961] http.c: match request [ari/channels/213075] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13961] http.c: match request [ari/channels/213075] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Announcer/ARI-00000026;2: Not in SFU mode [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is joining softmix technology [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: SIP/zvonobot-0000002a: [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: SIP/zvonobot-0000002a: [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: SIP/zvonobot-0000002a: Not in SFU mode [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is joining softmix technology [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Recorder/ARI-00000020;2: [Aug 18 10:34:04] DEBUG[13944] channel.c: Channel Recorder/ARI-00000020;2 setting write format path: slin -> slin [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:04] DEBUG[13944] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[13953] dial.c: Called zvonobot/79821116969 [Aug 18 10:34:04] DEBUG[13967] http.c: match request [ari/channels/213077] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13967] http.c: match request [ari/channels/213077] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13967] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13973] http.c: HTTP Request URI is /ari/channels/213079?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116961&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13961] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13972] http.c: match request [ari/channels/213078] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13972] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 640, ms is 60 [Aug 18 10:34:04] DEBUG[13973] http.c: match request [ari/channels/213079] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13961] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13975] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13973] http.c: match request [ari/channels/213079] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13973] http.c: match request [ari/channels/213079] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Finding handler for channels/213075 [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Recorder/ARI-00000020;2: [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13972] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[13975] http.c: HTTP Request URI is /ari/channels/213082?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116958&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13973] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Finding handler for channels/213078 [Aug 18 10:34:04] DEBUG[13947] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: pushing 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13973] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Finding handler for 213078 [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking channels create: Didn't match 213078 [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13972] res_ari.c: Checking channels externalMedia: Didn't match 213078 [Aug 18 10:34:04] DEBUG[13972] res_ari.c: No explicit handler found for 213078. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13976] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13944] bridge_softmix.c: Recorder/ARI-00000020;2: Not in SFU mode [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: Allocating new SIP dialog for 62e864aa2855afa70af7be9b3648ac1f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13957] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9408df40' [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) RTP allocated port 17578 [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE creating session 0.0.0.0:17578 (17578) [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE create [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE add system candidates [Aug 18 10:34:04] DEBUG[13957] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13957] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE add candidate: 159.65.48.104:17578, 2130706431 [Aug 18 10:34:04] DEBUG[13957] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13957] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE add candidate: 10.131.0.10:17578, 2130706431 [Aug 18 10:34:04] DEBUG[13957] rtp_engine.c: RTP instance '0x7f0c9408df40' is setup and ready to go [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) ICE stopped [Aug 18 10:34:04] DEBUG[13957] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13957] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13967] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:34:04] DEBUG[13975] http.c: match request [ari/channels/213082] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling softmix technology start [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13976] http.c: HTTP Request URI is /ari/channels/213080?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116960&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology destructor [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Finding handler for 213075 [Aug 18 10:34:04] DEBUG[13957] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13957] res_rtp_asterisk.c: (0x7f0c9408df40) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13957] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13957] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13957] chan_sip.c: SIP call-id changed from '62e864aa2855afa70af7be9b3648ac1f@127.0.1.1:5060' to '0c08b1570fa732272364833678dc04bb@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13957] stasis.c: Creating topic. name: channel:213076, detail: [Aug 18 10:34:04] DEBUG[13957] stasis.c: Topic 'channel:213076': 0x7f0c9409d400 created [Aug 18 10:34:04] DEBUG[13957] stasis.c: Creating topic. name: cache:280/channel:213076, detail: [Aug 18 10:34:04] DEBUG[13957] stasis.c: Topic 'cache:280/channel:213076': 0x7f0c9409de80 created [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Finding handler for channels/213079 [Aug 18 10:34:04] DEBUG[13976] http.c: match request [ari/channels/213080] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13978] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Finding handler for channels/213077 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking channels create: Didn't match 213075 [Aug 18 10:34:04] DEBUG[13976] http.c: match request [ari/channels/213080] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13975] http.c: match request [ari/channels/213082] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13961] res_ari.c: Checking channels externalMedia: Didn't match 213075 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13976] http.c: match request [ari/channels/213080] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:04] DEBUG[13978] http.c: HTTP Request URI is /ari/channels/213081?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116959&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13978] http.c: match request [ari/channels/213081] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13976] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13961] res_ari.c: No explicit handler found for 213075. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13975] http.c: match request [ari/channels/213082] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13979] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] http.c: HTTP Request URI is /ari/channels/213083?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116957&callerId=74950493843 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13976] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] http.c: match request [ari/channels/213083] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:04] DEBUG[13854] channel.c: Channel 0x7f0c38099400 'SIP/zvonobot-0000006d' allocated [Aug 18 10:34:04] DEBUG[13979] http.c: match request [ari/channels/213083] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:04] DEBUG[13854] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Finding handler for 213079 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking channels create: Didn't match 213079 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13973] res_ari.c: Checking channels externalMedia: Didn't match 213079 [Aug 18 10:34:04] DEBUG[13973] res_ari.c: No explicit handler found for 213079. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13975] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13979] http.c: match request [ari/channels/213083] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13978] http.c: match request [ari/channels/213081] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13978] http.c: match request [ari/channels/213081] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] http.c: Match made with [ari] [Aug 18 10:34:04] VERBOSE[13947] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:04] DEBUG[13775] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: Allocating new SIP dialog for 3f45e2785ceaba5c29e31f2a42740c2a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13951] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c0381d0' [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) RTP allocated port 10288 [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE creating session 0.0.0.0:10288 (10288) [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE create [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE add system candidates [Aug 18 10:34:04] DEBUG[13951] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13951] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE add candidate: 159.65.48.104:10288, 2130706431 [Aug 18 10:34:04] DEBUG[13951] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13951] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE add candidate: 10.131.0.10:10288, 2130706431 [Aug 18 10:34:04] DEBUG[13951] rtp_engine.c: RTP instance '0x7f0c8c0381d0' is setup and ready to go [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE stopped [Aug 18 10:34:04] DEBUG[13951] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13951] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13951] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13951] res_rtp_asterisk.c: (0x7f0c8c0381d0) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13951] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13951] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13975] http.c: HTTP consuming request body [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Finding handler for channels/213080 [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Finding handler for channels/213082 [Aug 18 10:34:04] DEBUG[13979] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13854] res_stasis.c: calls_0: Subscribing to 213072 [Aug 18 10:34:04] DEBUG[13854] stasis/app.c: Channel '213072' is 1 interested in calls_0 [Aug 18 10:34:04] DEBUG[13854] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag [Aug 18 10:34:04] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 - start 1629282843.387519 answer 1629282843.557781 end 1629282844.322772 dur 0.935 bill 0.764 dispo ANSWERED [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13854] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: Allocating new SIP dialog for 025459657dc0e6cc30061ec40835ceba@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13973] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca006da80' [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) RTP allocated port 17664 [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE creating session 0.0.0.0:17664 (17664) [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE create [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE add system candidates [Aug 18 10:34:04] DEBUG[13973] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13973] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE add candidate: 159.65.48.104:17664, 2130706431 [Aug 18 10:34:04] DEBUG[13973] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13973] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE add candidate: 10.131.0.10:17664, 2130706431 [Aug 18 10:34:04] DEBUG[13973] rtp_engine.c: RTP instance '0x7f0ca006da80' is setup and ready to go [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) ICE stopped [Aug 18 10:34:04] DEBUG[13973] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13973] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13973] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13973] res_rtp_asterisk.c: (0x7f0ca006da80) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13973] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13973] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13973] chan_sip.c: SIP call-id changed from '025459657dc0e6cc30061ec40835ceba@127.0.1.1:5060' to '40cf8f1449337acf099d514511a8313d@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13973] stasis.c: Creating topic. name: channel:213079, detail: [Aug 18 10:34:04] DEBUG[13973] stasis.c: Topic 'channel:213079': 0x7f0ca00ef730 created [Aug 18 10:34:04] DEBUG[13973] stasis.c: Creating topic. name: cache:281/channel:213079, detail: [Aug 18 10:34:04] DEBUG[13973] stasis.c: Topic 'cache:281/channel:213079': 0x7f0ca00f01b0 created [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: Allocating new SIP dialog for 24ab39412c0ce5e244ce8cb35adcfe6c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Outgoing Call for 79821116968 [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13978] http.c: HTTP consuming request body [Aug 18 10:34:04] DEBUG[13972] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9804b2b0' [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) RTP allocated port 10498 [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE creating session 0.0.0.0:10498 (10498) [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE create [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE add system candidates [Aug 18 10:34:04] DEBUG[13972] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13972] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE add candidate: 159.65.48.104:10498, 2130706431 [Aug 18 10:34:04] DEBUG[13972] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13972] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE add candidate: 10.131.0.10:10498, 2130706431 [Aug 18 10:34:04] DEBUG[13972] rtp_engine.c: RTP instance '0x7f0c9804b2b0' is setup and ready to go [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE stopped [Aug 18 10:34:04] DEBUG[13972] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13972] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13972] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13972] res_rtp_asterisk.c: (0x7f0c9804b2b0) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13972] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13764] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13972] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13972] chan_sip.c: SIP call-id changed from '24ab39412c0ce5e244ce8cb35adcfe6c@127.0.1.1:5060' to '3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13972] stasis.c: Creating topic. name: channel:213078, detail: [Aug 18 10:34:04] DEBUG[13972] stasis.c: Topic 'channel:213078': 0x7f0c9803f7c0 created [Aug 18 10:34:04] DEBUG[13972] stasis.c: Creating topic. name: cache:282/channel:213078, detail: [Aug 18 10:34:04] DEBUG[13972] stasis.c: Topic 'cache:282/channel:213078': 0x7f0c98040240 created [Aug 18 10:34:04] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Finding handler for channels/213081 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13846] res_stasis_playback.c: 1629282842.214: Sending play(sound:silence/2) command [Aug 18 10:34:04] DEBUG[13958] bridge_softmix.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: starting mixing thread [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13846] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:04] DEBUG[13846] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13947] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f'. Checking compatability for channels 'Snoop/213007-0000000d' and 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00' [Aug 18 10:34:04] DEBUG[13947] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' can not use native RTP bridge as could not get details [Aug 18 10:34:04] DEBUG[13947] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:04] DEBUG[13947] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:04] DEBUG[13947] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:04] DEBUG[13947] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:04] DEBUG[13947] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f is already using the new technology. [Aug 18 10:34:04] DEBUG[13947] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) is joining simple_bridge technology [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 setting read format path: slin16 -> slin16 [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel Snoop/213007-0000000d setting write format path: slin16 -> slin [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel Snoop/213007-0000000d setting read format path: slin -> slin16 [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 setting write format path: slin16 -> slin16 [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Finding handler for channels/213083 [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13951] chan_sip.c: SIP call-id changed from '3f45e2785ceaba5c29e31f2a42740c2a@127.0.1.1:5060' to '42462bcf58720fbb2059b6de455547db@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #14 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #14)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116984@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef Max-Forwards: 70 From: ;tag=as000dc064 To: Contact: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1791498542 1791498542 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19866 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13951] stasis.c: Creating topic. name: channel:213074, detail: [Aug 18 10:34:04] DEBUG[13951] stasis.c: Topic 'channel:213074': 0x7f0c8c05c950 created [Aug 18 10:34:04] DEBUG[13951] stasis.c: Creating topic. name: cache:283/channel:213074, detail: [Aug 18 10:34:04] DEBUG[13951] stasis.c: Topic 'cache:283/channel:213074': 0x7f0c8c0e7690 created [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13666] audiohook.c: Audiohook 0x7f0c80063290 has stale audio in its factories. Flushing them both [Aug 18 10:34:04] DEBUG[13666] res_rtp_asterisk.c: (0x7f0c74010590) RTP ooh, format changed from none to ulaw [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #31 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Finding handler for 213077 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking channels create: Didn't match 213077 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13967] res_ari.c: Checking channels externalMedia: Didn't match 213077 [Aug 18 10:34:04] DEBUG[13967] res_ari.c: No explicit handler found for 213077. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Finding handler for 213082 [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #31)) [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:04] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking channels create: Didn't match 213082 [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Finding handler for 213080 [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking channels create: Didn't match 213080 [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13976] res_ari.c: Checking channels externalMedia: Didn't match 213080 [Aug 18 10:34:04] DEBUG[13976] res_ari.c: No explicit handler found for 213080. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 736, ms is 112 [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #92 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #92)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116986@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7 Max-Forwards: 70 From: ;tag=as3da39e97 To: Contact: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168198193 1168198193 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13975] res_ari.c: Checking channels externalMedia: Didn't match 213082 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] VERBOSE[13695] res_rtp_asterisk.c: 0x7f0ca8020ff0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:16540 [Aug 18 10:34:04] VERBOSE[13981] chan_sip.c: Audio is at 11106 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 From: ;tag=as52d5bd88 To: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: *** SIP TIMER: Cancelling retransmission #63 - INVITE (got response) [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13975] res_ari.c: No explicit handler found for 213082. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: Allocating new SIP dialog for 0e5ca7c0754271cf5a84a84c6fcd37e6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13961] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca8034280' [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) RTP allocated port 16396 [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE creating session 0.0.0.0:16396 (16396) [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE create [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE add system candidates [Aug 18 10:34:04] DEBUG[13961] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13961] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE add candidate: 159.65.48.104:16396, 2130706431 [Aug 18 10:34:04] DEBUG[13961] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13961] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE add candidate: 10.131.0.10:16396, 2130706431 [Aug 18 10:34:04] DEBUG[13961] rtp_engine.c: RTP instance '0x7f0ca8034280' is setup and ready to go [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) ICE stopped [Aug 18 10:34:04] DEBUG[13961] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13961] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13961] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13961] res_rtp_asterisk.c: (0x7f0ca8034280) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13961] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13961] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] VERBOSE[13981] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13961] chan_sip.c: SIP call-id changed from '0e5ca7c0754271cf5a84a84c6fcd37e6@127.0.1.1:5060' to '6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] VERBOSE[13981] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Finding handler for 213083 [Aug 18 10:34:04] DEBUG[13982] channel.c: Channel Announcer/ARI-00000026;1 setting write format path: gsm -> slin [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 76 bytes from 178.62.121.41:16541 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Finding handler for 213081 [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking channels create: Didn't match 213083 [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking channels create: Didn't match 213081 [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Destroying SIP dialog 718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '718ba60c2d7ac569108f47132c014e8d@159.65.48.104:5060' Method: BYE [Aug 18 10:34:04] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS stop [Aug 18 10:34:04] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:04] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:04] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8001c6f0) ICE RTP transport deallocating [Aug 18 10:34:04] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8001c6f0' [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #92 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #92)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (2) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116981@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0 Max-Forwards: 70 From: ;tag=as76ba9cfd To: Contact: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1486321990 1486321990 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #17 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #17)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116983@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2 Max-Forwards: 70 From: ;tag=as00c25c39 To: Contact: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 610643329 610643329 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13961] stasis.c: Creating topic. name: channel:213075, detail: [Aug 18 10:34:04] DEBUG[13978] res_ari.c: Checking channels externalMedia: Didn't match 213081 [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13982] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:04] VERBOSE[13982] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[13981] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13979] res_ari.c: Checking channels externalMedia: Didn't match 213083 [Aug 18 10:34:04] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] res_ari.c: No explicit handler found for 213081. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13979] res_ari.c: No explicit handler found for 213083. Using wildcard channelId. [Aug 18 10:34:04] DEBUG[13958] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 688, ms is 63 [Aug 18 10:34:04] DEBUG[13905] stasis/app.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' is 4 interested in calls_0 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:04] DEBUG[13933] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:04] DEBUG[13933] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13958] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13984] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:04] DEBUG[13947] channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 setting write format path: slin -> slin16 [Aug 18 10:34:04] DEBUG[13947] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP ooh, format changed from none to slin16 [Aug 18 10:34:04] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 29 instead [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13984] http.c: HTTP Request URI is /ari/playbacks/20b8a8fd-704b-4feb-a6e5-05852658ad84 [Aug 18 10:34:04] DEBUG[13961] stasis.c: Topic 'channel:213075': 0x7f0ca806b390 created [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 From: ;tag=as6d7500c6 To: ;tag=as14883ee4 Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Initializing initreq for method INVITE - callid 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13984] http.c: match request [ari/playbacks/20b8a8fd-704b-4feb-a6e5-05852658ad84] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13961] stasis.c: Creating topic. name: cache:284/channel:213075, detail: [Aug 18 10:34:04] DEBUG[13961] stasis.c: Topic 'cache:284/channel:213075': 0x7f0ca807a110 created [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116968@178.62.121.41 SIP/2.0 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 3 [ 52]: From: ;tag=as3a3fa466 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 6 [ 60]: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:04 GMT [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:04] VERBOSE[13981] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #86 [Aug 18 10:34:04] DEBUG[13981] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6d7500c6 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as14883ee4 [Aug 18 10:34:04] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 1152, ms is 92 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13984] http.c: match request [ari/playbacks/20b8a8fd-704b-4feb-a6e5-05852658ad84] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:04] VERBOSE[13704] res_rtp_asterisk.c: 0x7f0c1c022950 -- Strict RTP learning complete - Locking on source address 178.62.121.41:18112 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[13981] dial.c: Called zvonobot/79821116968 [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13984] http.c: match request [ari/playbacks/20b8a8fd-704b-4feb-a6e5-05852658ad84] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: Allocating new SIP dialog for 26737bbd4f593de82f4825325dd68ae6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13978] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac077690' [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) RTP allocated port 11206 [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE creating session 0.0.0.0:11206 (11206) [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE create [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE add system candidates [Aug 18 10:34:04] DEBUG[13978] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13978] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE add candidate: 159.65.48.104:11206, 2130706431 [Aug 18 10:34:04] DEBUG[13978] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13978] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE add candidate: 10.131.0.10:11206, 2130706431 [Aug 18 10:34:04] DEBUG[13978] rtp_engine.c: RTP instance '0x7f0cac077690' is setup and ready to go [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) ICE stopped [Aug 18 10:34:04] DEBUG[13978] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13978] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 (Checking To) --From tag as6d7500c6 --To-tag as14883ee4 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:04] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP got report of 76 bytes from 178.62.121.41:18113 [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: Allocating new SIP dialog for 0adc17de5b683bbf1f8944da7d104aab@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13975] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca402ac60' [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) RTP allocated port 11576 [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE creating session 0.0.0.0:11576 (11576) [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE create [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE add system candidates [Aug 18 10:34:04] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 832, ms is 72 [Aug 18 10:34:04] DEBUG[13984] http.c: Match made with [ari] [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #64 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Stopping retransmission on '54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:04] DEBUG[13152] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13975] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9 Max-Forwards: 70 From: ;tag=as6d7500c6 To: ;tag=as14883ee4 Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:04] DEBUG[13975] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE add candidate: 159.65.48.104:11576, 2130706431 [Aug 18 10:34:04] DEBUG[13975] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13975] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE add candidate: 10.131.0.10:11576, 2130706431 [Aug 18 10:34:04] DEBUG[13975] rtp_engine.c: RTP instance '0x7f0ca402ac60' is setup and ready to go [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE stopped [Aug 18 10:34:04] DEBUG[13975] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13975] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13975] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13975] res_rtp_asterisk.c: (0x7f0ca402ac60) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13975] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13975] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13975] chan_sip.c: SIP call-id changed from '0adc17de5b683bbf1f8944da7d104aab@127.0.1.1:5060' to '4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13975] stasis.c: Creating topic. name: channel:213082, detail: [Aug 18 10:34:04] DEBUG[13975] stasis.c: Topic 'channel:213082': 0x7f0ca4042b80 created [Aug 18 10:34:04] DEBUG[13975] stasis.c: Creating topic. name: cache:285/channel:213082, detail: [Aug 18 10:34:04] DEBUG[13975] stasis.c: Topic 'cache:285/channel:213082': 0x7f0ca4043600 created [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Finding handler for playbacks/20b8a8fd-704b-4feb-a6e5-05852658ad84 [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Finding handler for playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Finding handler for 20b8a8fd-704b-4feb-a6e5-05852658ad84 [Aug 18 10:34:04] DEBUG[13984] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:04] DEBUG[13984] res_ari.c: No explicit handler found for 20b8a8fd-704b-4feb-a6e5-05852658ad84. Using wildcard playbackId. [Aug 18 10:34:04] DEBUG[13984] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:04] DEBUG[13978] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13984] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Audio is at 13804 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #61 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 640, ms is 60 [Aug 18 10:34:04] DEBUG[13992] http.c: HTTP opening session. Top level [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[13153] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13992] http.c: HTTP Request URI is /ari/channels/robot_212964 [Aug 18 10:34:04] DEBUG[13978] res_rtp_asterisk.c: (0x7f0cac077690) RTCP setup on RTP instance [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116985@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001 Max-Forwards: 70 From: ;tag=as697b28a1 To: Contact: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 78473575 78473575 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 672, ms is 62 [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:04] DEBUG[13992] http.c: match request [ari/channels/robot_212964] with handler [httpstatus] len 10 [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[13992] http.c: match request [ari/channels/robot_212964] with handler [phoneprov] len 9 [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:04] DEBUG[13879] channel.c: Channel 0x7f0c980440c0 'Announcer/ARI-00000027;1' allocated [Aug 18 10:34:04] DEBUG[13879] stasis.c: Creating topic. name: channel:1629282844.244, detail: [Aug 18 10:34:04] DEBUG[13879] stasis.c: Topic 'channel:1629282844.244': 0x7f0c980aceb0 created [Aug 18 10:34:04] DEBUG[13879] stasis.c: Creating topic. name: cache:286/channel:1629282844.244, detail: [Aug 18 10:34:04] DEBUG[13879] stasis.c: Topic 'cache:286/channel:1629282844.244': 0x7f0c980ad080 created [Aug 18 10:34:04] DEBUG[13992] http.c: match request [ari/channels/robot_212964] with handler [ari] len 3 [Aug 18 10:34:04] DEBUG[13992] http.c: Match made with [ari] [Aug 18 10:34:04] VERBOSE[13978] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13978] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13153] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:04] DEBUG[13153] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:04] DEBUG[13153] channel.c: Channel Announcer/ARI-00000002;1 setting write format path: slin -> slin [Aug 18 10:34:04] NOTICE[13153] res_stasis_playback.c: 1629282829.48: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:04] DEBUG[13153] channel.c: Channel 0x7f0c20014960 'Announcer/ARI-00000002;1' hanging up. Refs: 2 [Aug 18 10:34:04] DEBUG[13976] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Finding handler for channels/robot_212964 [Aug 18 10:34:04] DEBUG[13764] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 From: ;tag=as3d872a68 To: ;tag=as578d3717 Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as578d3717 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag as578d3717 [Aug 18 10:34:04] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Finding handler for channels [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: Allocating new SIP dialog for 7a41c020461b72fa0c0f60526ff37ecc@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13967] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c0840e0' [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) RTP allocated port 14548 [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE creating session 0.0.0.0:14548 (14548) [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE create [Aug 18 10:34:04] DEBUG[13764] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:04] DEBUG[13976] chan_sip.c: Allocating new SIP dialog for 533aafa20c63f5e75ea30b093c80aed8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13978] chan_sip.c: SIP call-id changed from '26737bbd4f593de82f4825325dd68ae6@127.0.1.1:5060' to '77ec81a43645f30730cc74c217742e98@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #57 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Stopping retransmission on '7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077 Max-Forwards: 70 From: ;tag=as3d872a68 To: ;tag=as578d3717 Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Audio is at 10086 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13882] channel.c: Channel 0x7f0ca00dd400 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80' allocated [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[13553] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13882] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Aug 18 10:34:04] DEBUG[13976] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb01036b0' [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) RTP allocated port 15846 [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE creating session 0.0.0.0:15846 (15846) [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE create [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:04] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[13978] stasis.c: Creating topic. name: channel:213081, detail: [Aug 18 10:34:04] DEBUG[13978] stasis.c: Topic 'channel:213081': 0x7f0cac097940 created [Aug 18 10:34:04] DEBUG[13978] stasis.c: Creating topic. name: cache:287/channel:213081, detail: [Aug 18 10:34:04] DEBUG[13978] stasis.c: Topic 'cache:287/channel:213081': 0x7f0cac098340 created [Aug 18 10:34:04] VERBOSE[13882] res_rtp_asterisk.c: 0x7f0ca0027810 -- Strict RTP learning after remote address set to: 127.0.0.1:50264 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE add system candidates [Aug 18 10:34:04] DEBUG[13967] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13967] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE add system candidates [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE add candidate: 159.65.48.104:14548, 2130706431 [Aug 18 10:34:04] DEBUG[13967] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13967] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE add candidate: 10.131.0.10:14548, 2130706431 [Aug 18 10:34:04] DEBUG[13967] rtp_engine.c: RTP instance '0x7f0c9c0840e0' is setup and ready to go [Aug 18 10:34:04] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP audio difference is 680, ms is 105 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #70 (5) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #70)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116982@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4 Max-Forwards: 70 From: ;tag=as08a5ad00 To: Contact: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 904249084 904249084 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11586 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #92 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #92)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (3) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #31 (4) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #31)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (1) INVITE - 5 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13976] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:04] DEBUG[13882] res_stasis.c: calls_0: Subscribing to robot_213009 [Aug 18 10:34:04] DEBUG[13882] stasis/app.c: Channel 'robot_213009' is 1 interested in calls_0 [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:04] DEBUG[13882] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:04] DEBUG[13882] http.c: HTTP closing session. Top level [Aug 18 10:34:04] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13764] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as0a75a671 Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a75a671 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as0a75a671 [Aug 18 10:34:04] DEBUG[13976] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE stopped [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE add candidate: 159.65.48.104:15846, 2130706431 [Aug 18 10:34:04] DEBUG[13976] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:04] DEBUG[13976] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:04] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 768, ms is 116 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Stopping retransmission on '16541045053beef31ed8e5361337aa22@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:04] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE add candidate: 10.131.0.10:15846, 2130706431 [Aug 18 10:34:04] DEBUG[13967] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:04] DEBUG[13967] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:04] DEBUG[13967] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:04] DEBUG[13967] res_rtp_asterisk.c: (0x7f0c9c0840e0) RTCP setup on RTP instance [Aug 18 10:34:04] VERBOSE[13967] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:04] DEBUG[13967] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:04] DEBUG[13967] chan_sip.c: SIP call-id changed from '7a41c020461b72fa0c0f60526ff37ecc@127.0.1.1:5060' to '564726e17074235c1af6801638e43e42@159.65.48.104:5060' [Aug 18 10:34:04] DEBUG[13967] stasis.c: Creating topic. name: channel:213077, detail: [Aug 18 10:34:04] DEBUG[13967] stasis.c: Topic 'channel:213077': 0x7f0c9c08f170 created [Aug 18 10:34:04] DEBUG[13967] stasis.c: Creating topic. name: cache:288/channel:213077, detail: [Aug 18 10:34:04] DEBUG[13967] stasis.c: Topic 'cache:288/channel:213077': 0x7f0c9c08fc60 created [Aug 18 10:34:04] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:04] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:04] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:04] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Finding handler for robot_212964 [Aug 18 10:34:04] DEBUG[13979] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:04] DEBUG[13979] chan_sip.c: Allocating new SIP dialog for 0eff33ba01cc263b211eede648807617@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:04] DEBUG[13979] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb408bdf0' [Aug 18 10:34:04] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) RTP allocated port 18420 [Aug 18 10:34:04] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE creating session 0.0.0.0:18420 (18420) [Aug 18 10:34:04] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE create [Aug 18 10:34:04] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE add system candidates [Aug 18 10:34:04] DEBUG[13979] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:04] DEBUG[13979] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 From: ;tag=as63ca65c0 To: ;tag=as56bda81b Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" Content-Length: 0 <-------------> [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56bda81b [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:04] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:04] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag as56bda81b [Aug 18 10:34:04] DEBUG[13976] rtp_engine.c: RTP instance '0x7f0cb01036b0' is setup and ready to go [Aug 18 10:34:05] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE add candidate: 159.65.48.104:18420, 2130706431 [Aug 18 10:34:05] DEBUG[13979] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:05] DEBUG[13979] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:05] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE add candidate: 10.131.0.10:18420, 2130706431 [Aug 18 10:34:05] DEBUG[13979] rtp_engine.c: RTP instance '0x7f0cb408bdf0' is setup and ready to go [Aug 18 10:34:05] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE stopped [Aug 18 10:34:05] DEBUG[13979] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:05] DEBUG[13979] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:05] DEBUG[13979] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:05] DEBUG[13979] res_rtp_asterisk.c: (0x7f0cb408bdf0) RTCP setup on RTP instance [Aug 18 10:34:04] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '63c7430f0c6f8039194559375e330124@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:04] DEBUG[13992] res_ari.c: Checking channels create: Didn't match robot_212964 [Aug 18 10:34:05] VERBOSE[13979] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:05] DEBUG[13979] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:05] DEBUG[13979] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (2) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (1) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 From: ;tag=as52d5bd88 To: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag [Aug 18 10:34:05] DEBUG[13979] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:05] DEBUG[13979] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13979] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:05] DEBUG[13979] chan_sip.c: SIP call-id changed from '0eff33ba01cc263b211eede648807617@127.0.1.1:5060' to '4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060' [Aug 18 10:34:05] DEBUG[13979] stasis.c: Creating topic. name: channel:213083, detail: [Aug 18 10:34:05] DEBUG[13979] stasis.c: Topic 'channel:213083': 0x7f0cb4082b30 created [Aug 18 10:34:05] DEBUG[13979] stasis.c: Creating topic. name: cache:289/channel:213083, detail: [Aug 18 10:34:05] DEBUG[13979] stasis.c: Topic 'cache:289/channel:213083': 0x7f0cb4083570 created [Aug 18 10:34:05] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE stopped [Aug 18 10:34:05] DEBUG[13976] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:05] DEBUG[13976] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:05] DEBUG[13976] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:05] DEBUG[13976] res_rtp_asterisk.c: (0x7f0cb01036b0) RTCP setup on RTP instance [Aug 18 10:34:05] VERBOSE[13976] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:05] DEBUG[13976] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:05] DEBUG[13976] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:05] DEBUG[13976] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:05] DEBUG[13976] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13976] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:05] DEBUG[13976] chan_sip.c: SIP call-id changed from '533aafa20c63f5e75ea30b093c80aed8@127.0.1.1:5060' to '467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060' [Aug 18 10:34:05] DEBUG[13976] stasis.c: Creating topic. name: channel:213080, detail: [Aug 18 10:34:05] DEBUG[13992] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13976] stasis.c: Topic 'channel:213080': 0x7f0cb0162c50 created [Aug 18 10:34:05] DEBUG[13976] stasis.c: Creating topic. name: cache:290/channel:213080, detail: [Aug 18 10:34:05] DEBUG[13976] stasis.c: Topic 'cache:290/channel:213080': 0x7f0cb0058100 created [Aug 18 10:34:05] VERBOSE[13993] dial.c: Called 127.0.0.1:50264 [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13992] res_ari.c: Checking channels externalMedia: Didn't match robot_212964 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:05] DEBUG[13900] channel.c: Channel 0x7f0c1c127e20 'Announcer/ARI-00000028;1' allocated [Aug 18 10:34:05] DEBUG[13900] stasis.c: Creating topic. name: channel:1629282845.249, detail: [Aug 18 10:34:05] DEBUG[13900] stasis.c: Topic 'channel:1629282845.249': 0x7f0c1c13fb50 created [Aug 18 10:34:05] DEBUG[13900] stasis.c: Creating topic. name: cache:291/channel:1629282845.249, detail: [Aug 18 10:34:05] DEBUG[13900] stasis.c: Topic 'cache:291/channel:1629282845.249': 0x7f0c1c140580 created [Aug 18 10:34:05] DEBUG[13992] res_ari.c: No explicit handler found for robot_212964. Using wildcard channelId. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (5) INVITE - 5 [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:05] DEBUG[13840] channel.c: Channel 0x7f0c2c0c8bd0 'SIP/zvonobot-0000006a' allocated [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13840] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116978@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0 Max-Forwards: 70 From: ;tag=as080d6dff To: Contact: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 808826032 808826032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17282 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (5) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116980@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7c2cb0aa Max-Forwards: 70 From: ;tag=as73898a35 To: Contact: Call-ID: 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 422140348 422140348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12964 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13840] res_stasis.c: calls_0: Subscribing to 213070 [Aug 18 10:34:05] DEBUG[13840] stasis/app.c: Channel '213070' is 1 interested in calls_0 [Aug 18 10:34:05] VERBOSE[13993] dial.c: UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 answered [Aug 18 10:34:05] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13840] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13840] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Outgoing Call for 79821116970 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[13994] chan_sip.c: Audio is at 14444 [Aug 18 10:34:05] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] VERBOSE[13994] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 688, ms is 63 [Aug 18 10:34:05] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:05] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:05] VERBOSE[13993] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 [Aug 18 10:34:05] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 From: ;tag=as1dc5ead1 To: ;tag=as741846ee Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1dc5ead1 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as741846ee [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" [Aug 18 10:34:05] VERBOSE[13994] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 (Checking To) --From tag as1dc5ead1 --To-tag as741846ee [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (2) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (5) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116992@178.62.121.41", nonce="5bd640e9", response="7310ac6ba0055b9525729e1647a09ccf" Date: Wed, 18 Aug 2021 10:34:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 299638690 299638691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18824 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (5) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116993@178.62.121.41", nonce="1dc16f41", response="686f15f3bb4b0d2da8301571d1c636ec" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698982166 698982167 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (5) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116977@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9 Max-Forwards: 70 From: ;tag=as6ac21020 To: Contact: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1439110535 1439110535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16058 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (2) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 760, ms is 115 [Aug 18 10:34:05] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 From: ;tag=as02885f54 To: ;tag=as5580464d Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as02885f54 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5580464d [Aug 18 10:34:05] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13996] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[13996] http.c: HTTP Request URI is /ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:05] DEBUG[13996] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [httpstatus] len 10 [Aug 18 10:34:05] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 1040, ms is 85 [Aug 18 10:34:05] DEBUG[13996] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:05] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:05] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 (Checking To) --From tag as02885f54 --To-tag as5580464d [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 896, ms is 76 [Aug 18 10:34:05] DEBUG[13996] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[13996] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '10507dcf059680b46ad884550335c862@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (5) INVITE - 5 [Aug 18 10:34:05] VERBOSE[13994] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116979@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580 Max-Forwards: 70 From: ;tag=as54647e8b To: Contact: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1951127551 1951127551 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Initializing initreq for method INVITE - callid 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116970@178.62.121.41 SIP/2.0 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:05] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Finding handler for bridges/357a4882-a24d-489f-8ff8-98badd81b2ee/play [Aug 18 10:34:05] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP audio difference is 736, ms is 112 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 3 [ 52]: From: ;tag=as2eb39fa6 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 6 [ 60]: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP audio difference is 1248, ms is 176 [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Finding handler for bridges [Aug 18 10:34:05] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 From: ;tag=as0261f463 To: ;tag=as64b58d1a Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 460639390 460639390 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0261f463 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64b58d1a [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 460639390 460639390 IN IP4 178.62.121.41 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17848 RTP/AVP 0 8 101 [Aug 18 10:34:05] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13993] stasis/app.c: Channel 'robot_213009' is 2 interested in calls_0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking To) --From tag as0261f463 --To-tag as64b58d1a [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Strict routing enforced for session 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:05] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:05] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117049@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00adc8f5 Max-Forwards: 70 From: ;tag=as0261f463 To: ;tag=as64b58d1a Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] VERBOSE[13994] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Aug 18 10:34:05] DEBUG[13994] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Finding handler for 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:05] DEBUG[13996] res_ari.c: No explicit handler found for 357a4882-a24d-489f-8ff8-98badd81b2ee. Using wildcard bridgeId. [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Finding handler for play [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:05] VERBOSE[13994] dial.c: Called zvonobot/79821116970 [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[14005] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[14005] http.c: HTTP Request URI is /ari/bridges/051b3352-0990-44a6-b6a2-2bd678146686/addChannel?channel=1629282842.221%2Crobot_213009 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31c5c295 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag as31c5c295 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117048@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1 Max-Forwards: 70 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Contact: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:05] DEBUG[13996] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:05] DEBUG[14005] http.c: match request [ari/bridges/051b3352-0990-44a6-b6a2-2bd678146686/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:05] DEBUG[14005] http.c: match request [ari/bridges/051b3352-0990-44a6-b6a2-2bd678146686/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[14005] http.c: match request [ari/bridges/051b3352-0990-44a6-b6a2-2bd678146686/addChannel] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[14005] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Finding handler for bridges/051b3352-0990-44a6-b6a2-2bd678146686/addChannel [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Finding handler for bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Finding handler for 051b3352-0990-44a6-b6a2-2bd678146686 [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:05] DEBUG[14005] res_ari.c: No explicit handler found for 051b3352-0990-44a6-b6a2-2bd678146686. Using wildcard bridgeId. [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Finding handler for addChannel [Aug 18 10:34:05] DEBUG[14005] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:05] DEBUG[14005] stasis/control.c: 1629282842.221: Sending channel add_to_bridge command [Aug 18 10:34:05] DEBUG[13906] channel.c: Channel 0x7f0c240f6d50 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30' allocated [Aug 18 10:34:05] DEBUG[13906] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:05] VERBOSE[13906] res_rtp_asterisk.c: 0x7f0c240f2dc0 -- Strict RTP learning after remote address set to: 127.0.0.1:50349 [Aug 18 10:34:05] DEBUG[13906] res_stasis.c: calls_0: Subscribing to robot_213011 [Aug 18 10:34:05] DEBUG[13906] stasis/app.c: Channel 'robot_213011' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[13050] chan_sip.c: Hangup call SIP/zvonobot-0000001d, SIP callid 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13906] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13906] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[13770] channel.c: Channel 0x7f0c90063430 'Announcer/ARI-00000021;1' destroying [Aug 18 10:34:05] DEBUG[13760] bridge_channel.c: Setting 0x7f0c90058340(Announcer/ARI-00000021;2) state from:0 to:1 [Aug 18 10:34:05] DEBUG[13770] stasis.c: Destroying topic. name: cache:223/channel:1629282839.188, detail: [Aug 18 10:34:05] DEBUG[13050] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[13770] stasis.c: Topic 'cache:223/channel:1629282839.188': 0x7f0c9005c460 destroyed [Aug 18 10:34:05] DEBUG[13050] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[13050] channel.c: Channel 0x7f0c9001fe80 'SIP/zvonobot-0000001d' destroying [Aug 18 10:34:05] DEBUG[13761] bridge_channel.c: Setting 0x7f0c0804a470(Announcer/ARI-00000022;2) state from:0 to:1 [Aug 18 10:34:05] DEBUG[13553] bridge_channel.c: Setting 0x7f0cb4042750(Snoop/212999-00000008) state from:0 to:1 [Aug 18 10:34:05] DEBUG[13462] channel.c: Channel 0x7f0c7c017b90 'Recorder/ARI-00000013;2' destroying [Aug 18 10:34:05] DEBUG[13873] bridge_roles.c: Roles did not exist on channel Snoop/213009-0000000e [Aug 18 10:34:05] VERBOSE[13465] app.c: User hung up [Aug 18 10:34:05] DEBUG[13761] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pulling 0x7f0c0804a470(Announcer/ARI-00000022;2) [Aug 18 10:34:05] VERBOSE[13761] bridge_channel.c: Channel Announcer/ARI-00000022;2 left 'softmix' stasis-bridge [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:05] DEBUG[13761] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c0804a470(Announcer/ARI-00000022;2) is leaving softmix technology [Aug 18 10:34:05] DEBUG[13465] res_stasis_recording.c: 1629282835.128: Recording complete [Aug 18 10:34:05] DEBUG[13873] stasis/control.c: 1629282842.221: Adding to bridge 051b3352-0990-44a6-b6a2-2bd678146686 [Aug 18 10:34:05] DEBUG[13129] chan_sip.c: Hangup call SIP/zvonobot-00000023, SIP callid 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[13760] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pulling 0x7f0c90058340(Announcer/ARI-00000021;2) [Aug 18 10:34:05] VERBOSE[13760] bridge_channel.c: Channel Announcer/ARI-00000021;2 left 'softmix' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:34:05] DEBUG[13760] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c90058340(Announcer/ARI-00000021;2) is leaving softmix technology [Aug 18 10:34:05] DEBUG[13465] channel.c: Channel 0x7f0c7c00f100 'Recorder/ARI-00000013;1' hanging up. Refs: 2 [Aug 18 10:34:05] DEBUG[13553] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: pulling 0x7f0cb4042750(Snoop/212999-00000008) [Aug 18 10:34:05] VERBOSE[13553] bridge_channel.c: Channel Snoop/212999-00000008 left 'simple_bridge' stasis-bridge [Aug 18 10:34:05] DEBUG[13553] bridge_channel.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: 0x7f0cb4042750(Snoop/212999-00000008) is leaving simple_bridge technology [Aug 18 10:34:05] DEBUG[13778] channel.c: Channel 0x7f0c08044260 'Announcer/ARI-00000022;1' destroying [Aug 18 10:34:05] DEBUG[13770] stasis.c: Destroying topic. name: channel:1629282839.188, detail: [Aug 18 10:34:05] DEBUG[13129] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[13129] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[13873] stasis/app.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13761] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6'. Checking compatability for channels 'SIP/zvonobot-0000000e' and 'Recorder/ARI-00000019;2' [Aug 18 10:34:05] DEBUG[14015] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: 0x7f0c9c02a650(Snoop/213009-0000000e) is joining [Aug 18 10:34:05] DEBUG[14015] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: pushing 0x7f0c9c02a650(Snoop/213009-0000000e) [Aug 18 10:34:05] VERBOSE[14015] bridge_channel.c: Channel Snoop/213009-0000000e joined 'simple_bridge' stasis-bridge <051b3352-0990-44a6-b6a2-2bd678146686> [Aug 18 10:34:05] DEBUG[13761] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as channel 'SIP/zvonobot-0000000e' has features which prevent it [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: channel '212992': is 0 interested in calls_0 [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: channel '212992' unsubscribed from calls_0 [Aug 18 10:34:05] DEBUG[13770] stasis.c: Topic 'channel:1629282839.188': 0x7f0c90059200 destroyed [Aug 18 10:34:05] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282845.250, detail: [Aug 18 10:34:05] VERBOSE[13129] chan_sip.c: Scheduling destruction of SIP dialog '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' in 6400 ms (Method: INVITE) [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13553] bridge_native_rtp.c: Bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' can not use native RTP bridge as two channels are required [Aug 18 10:34:05] DEBUG[13553] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[13761] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] DEBUG[13553] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] VERBOSE[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: switching from softmix technology to simple_bridge [Aug 18 10:34:05] DEBUG[13553] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13553] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] DEBUG[13553] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3 is already using the new technology. [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology constructor [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: cache:36/channel:212992, detail: [Aug 18 10:34:05] DEBUG[20545] stasis.c: Topic 'channel:1629282845.250': 0x7f0c300ba000 created [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c9809c220(SIP/zvonobot-0000000e) to dummy bridge temporarily [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) to dummy bridge temporarily [Aug 18 10:34:05] DEBUG[13553] bridge_channel.c: Bridge is returning 0x7f0cb4042750(Snoop/212999-00000008) to read format slin [Aug 18 10:34:05] DEBUG[20545] stasis.c: Creating topic. name: cache:292/channel:1629282845.250, detail: [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is leaving softmix technology (dummy) [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is leaving softmix technology (dummy) [Aug 18 10:34:05] DEBUG[13553] channel.c: Channel Snoop/212999-00000008 setting read format path: slin -> slin [Aug 18 10:34:05] DEBUG[20545] stasis.c: Topic 'cache:292/channel:1629282845.250': 0x7f0c300b2f10 created [Aug 18 10:34:05] DEBUG[13553] bridge_channel.c: Bridge is returning 0x7f0cb4042750(Snoop/212999-00000008) to write format slin [Aug 18 10:34:05] DEBUG[13553] channel.c: Channel Snoop/212999-00000008 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[13553] stasis/control.c: 1629282835.132, c66c6480-4085-4bd9-87d2-ee6f5748dcc3: Channel was departed from bridge [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'cache:36/channel:212992': 0x7f0c90021460 destroyed [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology stop [Aug 18 10:34:05] DEBUG[13553] stasis/app.c: bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3': is 2 interested in calls_0 [Aug 18 10:34:05] DEBUG[13553] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:05] DEBUG[13474] stasis/control.c: 1629282835.132: Channel departing bridge [Aug 18 10:34:05] DEBUG[13474] bridge.c: Waiting for 0x7f0cb4042750(Snoop/212999-00000008) bridge thread to die. [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[13474] stasis/app.c: channel '1629282835.132': is 0 interested in calls_0 [Aug 18 10:34:05] DEBUG[13474] stasis/app.c: channel '1629282835.132' unsubscribed from calls_0 [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13462] stasis.c: Destroying topic. name: cache:154/channel:1629282835.129, detail: [Aug 18 10:34:05] DEBUG[13474] channel.c: Channel 0x7f0ca0056680 'Snoop/212999-00000008' hanging up. Refs: 3 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:05] VERBOSE[14011] dial.c: Called 127.0.0.1:50349 [Aug 18 10:34:05] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20545] stasis.c: Destroying topic. name: cache:292/channel:1629282845.250, detail: [Aug 18 10:34:05] DEBUG[20545] stasis.c: Topic 'cache:292/channel:1629282845.250': 0x7f0c300b2f10 destroyed [Aug 18 10:34:05] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282845.250, detail: [Aug 18 10:34:05] DEBUG[20545] stasis.c: Topic 'channel:1629282845.250': 0x7f0c300ba000 destroyed [Aug 18 10:34:05] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001d', '', 'AppDial2', '(Outgoing Line)', 17, 0, 'BUSY', 3, '', '212992', '')] [Aug 18 10:34:05] DEBUG[13761] channel.c: Channel Recorder/ARI-00000019;2 setting read format path: slin -> slin [Aug 18 10:34:05] DEBUG[13462] stasis.c: Topic 'cache:154/channel:1629282835.129': 0x7f0c7c016ec0 destroyed [Aug 18 10:34:05] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13129] chan_sip.c: Strict routing enforced for session 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:34:05] VERBOSE[13129] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:05] DEBUG[13129] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:05] DEBUG[13129] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:05] VERBOSE[13129] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[13129] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[13129] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #55 [Aug 18 10:34:05] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1024, ms is 84 [Aug 18 10:34:05] DEBUG[13462] stasis.c: Destroying topic. name: channel:1629282835.129, detail: [Aug 18 10:34:05] DEBUG[13462] stasis.c: Topic 'channel:1629282835.129': 0x7f0c7c047e00 destroyed [Aug 18 10:34:05] DEBUG[13761] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[13761] channel.c: Channel Recorder/ARI-00000019;2 setting read format path: slin -> slin [Aug 18 10:34:05] DEBUG[13761] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology start [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: deferring softmix technology destructor [Aug 18 10:34:05] DEBUG[13761] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: queueing action type:13 sub:1000 [Aug 18 10:34:05] DEBUG[13761] channel.c: Channel 0x7f0c0806c2f0 'Announcer/ARI-00000022;2' hanging up. Refs: 2 [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:05] DEBUG[20534] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: Waiting for mixing thread to die. [Aug 18 10:34:05] DEBUG[13775] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: stopping mixing thread [Aug 18 10:34:05] DEBUG[13760] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060'. Checking compatability for channels 'SIP/zvonobot-0000002b' and 'Recorder/ARI-0000001a;2' [Aug 18 10:34:05] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13129] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:05] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13760] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as channel 'SIP/zvonobot-0000002b' has features which prevent it [Aug 18 10:34:05] DEBUG[13624] channel.c: Recorder/ARI-00000019;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[13778] stasis.c: Destroying topic. name: cache:226/channel:1629282839.189, detail: [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13778] stasis.c: Topic 'cache:226/channel:1629282839.189': 0x7f0c0805c660 destroyed [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13760] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] VERBOSE[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: switching from softmix technology to simple_bridge [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 From: ;tag=as6093d024 To: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag [Aug 18 10:34:05] DEBUG[14015] bridge_native_rtp.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' can not use native RTP bridge as two channels are required [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology constructor [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c7803b960(SIP/zvonobot-0000002b) to dummy bridge temporarily [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) to dummy bridge temporarily [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is leaving softmix technology (dummy) [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is leaving softmix technology (dummy) [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology stop [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[13760] channel.c: Channel Recorder/ARI-0000001a;2 setting read format path: slin -> slin [Aug 18 10:34:05] DEBUG[13760] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[13760] channel.c: Channel Recorder/ARI-0000001a;2 setting read format path: slin -> slin [Aug 18 10:34:05] DEBUG[13760] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology start [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: deferring softmix technology destructor [Aug 18 10:34:05] DEBUG[13760] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: queueing action type:13 sub:1000 [Aug 18 10:34:05] DEBUG[13778] stasis.c: Destroying topic. name: channel:1629282839.189, detail: [Aug 18 10:34:05] DEBUG[13778] stasis.c: Topic 'channel:1629282839.189': 0x7f0c0807fe00 destroyed [Aug 18 10:34:05] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 688, ms is 63 [Aug 18 10:34:05] DEBUG[14015] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[14015] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 1056, ms is 86 [Aug 18 10:34:05] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP audio difference is 728, ms is 111 [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP audio difference is 912, ms is 134 [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: channel:212992, detail: [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'channel:212992': 0x7f0c900225d0 destroyed [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (3) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[14015] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[14015] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] DEBUG[14015] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686 is already using the new technology. [Aug 18 10:34:05] DEBUG[14015] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: 0x7f0c9c02a650(Snoop/213009-0000000e) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP audio difference is 704, ms is 108 [Aug 18 10:34:05] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP got report of 100 bytes from 178.62.121.41:10913 [Aug 18 10:34:05] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 712, ms is 109 [Aug 18 10:34:05] DEBUG[13888] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 1472, ms is 112 [Aug 18 10:34:05] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 760, ms is 115 [Aug 18 10:34:05] DEBUG[13764] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: stopping mixing thread [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:05] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTCP got report of 100 bytes from 178.62.121.41:10789 [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20534] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: Waiting for mixing thread to die. [Aug 18 10:34:05] DEBUG[13993] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 [Aug 18 10:34:05] DEBUG[13760] channel.c: Channel 0x7f0c90044400 'Announcer/ARI-00000021;2' hanging up. Refs: 2 [Aug 18 10:34:05] DEBUG[13626] channel.c: Recorder/ARI-0000001a;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[13873] stasis/app.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' is 2 interested in calls_0 [Aug 18 10:34:05] DEBUG[14005] stasis/control.c: robot_213009: Sending channel add_to_bridge command [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Session timer stopped: 30 - 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:34:05] VERBOSE[14011] dial.c: UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 answered [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (1) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 1008, ms is 83 [Aug 18 10:34:05] DEBUG[13556] channel.c: SIP/zvonobot-0000002b: Dropping redundant connected line update "" <>. [Aug 18 10:34:05] DEBUG[13936] channel.c: Soft-Hanging (0x20) up channel 'Snoop/212999-00000008' [Aug 18 10:34:05] DEBUG[13936] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:05] DEBUG[13550] channel.c: SIP/zvonobot-0000000e: Dropping redundant connected line update "" <>. [Aug 18 10:34:05] DEBUG[13936] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[13855] channel.c: Channel 0x7f0c400470e0 'SIP/zvonobot-00000068' allocated [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13855] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] DEBUG[14021] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[14021] http.c: HTTP Request URI is /ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:34:05] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:05] DEBUG[14021] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162] with handler [httpstatus] len 10 [Aug 18 10:34:05] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 10 instead [Aug 18 10:34:05] VERBOSE[14011] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 [Aug 18 10:34:05] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[14021] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[14021] http.c: match request [ari/bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[14021] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:05] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Finding handler for bridges/3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (3) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Destroying SIP dialog 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS stop [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c900183c0) ICE RTP transport deallocating [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c900183c0' [Aug 18 10:34:05] DEBUG[14011] stasis/app.c: Channel 'robot_213011' is 2 interested in calls_0 [Aug 18 10:34:05] DEBUG[13855] res_stasis.c: calls_0: Subscribing to 213073 [Aug 18 10:34:05] DEBUG[13855] stasis/app.c: Channel '213073' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[13855] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13855] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:05] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[13867] stasis.c: Creating topic. name: channel:1629282845.251, detail: [Aug 18 10:34:05] DEBUG[13867] stasis.c: Topic 'channel:1629282845.251': 0x7f0c8800f890 created [Aug 18 10:34:05] DEBUG[13867] stasis.c: Creating topic. name: cache:293/channel:1629282845.251, detail: [Aug 18 10:34:05] DEBUG[13867] stasis.c: Topic 'cache:293/channel:1629282845.251': 0x7f0c880756b0 created [Aug 18 10:34:05] DEBUG[13945] channel.c: Channel 0x7f0c78090610 'Recorder/ARI-00000029;1' allocated [Aug 18 10:34:05] DEBUG[13945] stasis.c: Creating topic. name: channel:1629282845.252, detail: [Aug 18 10:34:05] DEBUG[13945] stasis.c: Topic 'channel:1629282845.252': 0x7f0c7803c690 created [Aug 18 10:34:05] DEBUG[13945] stasis.c: Creating topic. name: cache:294/channel:1629282845.252, detail: [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Finding handler for bridges [Aug 18 10:34:05] DEBUG[13945] stasis.c: Topic 'cache:294/channel:1629282845.252': 0x7f0c7803cb70 created [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 From: ;tag=as52d5bd88 To: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag [Aug 18 10:34:05] DEBUG[13888] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:05] DEBUG[13888] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:05] DEBUG[13888] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:05] DEBUG[13888] channel.c: Channel Announcer/ARI-00000025;1 setting write format path: slin -> slin [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Outgoing Call for 79821116967 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[14035] chan_sip.c: Audio is at 18778 [Aug 18 10:34:05] VERBOSE[14035] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Finding handler for 3d1f9573-48d1-48a0-bcdd-9d7f09555162 [Aug 18 10:34:05] DEBUG[14021] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:05] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (1) BYE - 8 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[14021] res_ari.c: No explicit handler found for 3d1f9573-48d1-48a0-bcdd-9d7f09555162. Using wildcard bridgeId. [Aug 18 10:34:05] DEBUG[13957] channel.c: Channel 0x7f0c9409b680 'SIP/zvonobot-0000006e' allocated [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13957] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] VERBOSE[14035] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] DEBUG[14021] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: telling all channels to leave the party [Aug 18 10:34:05] DEBUG[14021] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:05] DEBUG[14021] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: queueing action type:13 sub:1001 [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling stasis bridge destructor [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology stop [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge 3d1f9573-48d1-48a0-bcdd-9d7f09555162: calling simple_bridge technology destructor [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162': is 0 interested in calls_0 [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: bridge '3d1f9573-48d1-48a0-bcdd-9d7f09555162' unsubscribed from calls_0 [Aug 18 10:34:05] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:05] DEBUG[14021] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:05] DEBUG[14021] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[14036] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: cache:144/bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162, detail: [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'cache:144/bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162': 0x7f0cb4045cd0 destroyed [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162, detail: [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'bridge:3d1f9573-48d1-48a0-bcdd-9d7f09555162': 0x7f0cb404aff0 destroyed [Aug 18 10:34:05] DEBUG[13888] channel.c: Channel 0x7f0c8408c060 'Announcer/ARI-00000025;1' hanging up. Refs: 2 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] VERBOSE[14035] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Initializing initreq for method INVITE - callid 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116967@178.62.121.41 SIP/2.0 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (3) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #92 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #92)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (2) INVITE - 5 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 3 [ 52]: From: ;tag=as46d7f260 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[14036] http.c: HTTP Request URI is /ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 6 [ 60]: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] VERBOSE[14035] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (4) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:05] DEBUG[13993] stasis/control.c: robot_213009: Adding to bridge 051b3352-0990-44a6-b6a2-2bd678146686 [Aug 18 10:34:05] DEBUG[13993] stasis/app.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' is 3 interested in calls_0 [Aug 18 10:34:05] DEBUG[14038] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) is joining [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[14038] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: pushing 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) [Aug 18 10:34:05] DEBUG[14035] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[14036] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3] with handler [httpstatus] len 10 [Aug 18 10:34:05] VERBOSE[14038] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 joined 'simple_bridge' stasis-bridge <051b3352-0990-44a6-b6a2-2bd678146686> [Aug 18 10:34:05] DEBUG[14036] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 - start 1629282844.903624 answer 1629282845.080418 end 1629282845.765408 dur 0.861 bill 0.684 dispo ANSWERED [Aug 18 10:34:05] DEBUG[14038] bridge_native_rtp.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686'. Checking compatability for channels 'Snoop/213009-0000000e' and 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80' [Aug 18 10:34:05] DEBUG[14036] http.c: match request [ari/bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[14038] bridge_native_rtp.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' can not use native RTP bridge as could not get details [Aug 18 10:34:05] DEBUG[14038] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[14038] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[14038] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[14038] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] DEBUG[14038] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686 is already using the new technology. [Aug 18 10:34:05] DEBUG[14038] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[14036] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[13957] res_stasis.c: calls_0: Subscribing to 213076 [Aug 18 10:34:05] DEBUG[13957] stasis/app.c: Channel '213076' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Outgoing Call for 79821116964 [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 setting read format path: slin16 -> slin16 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[13957] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13957] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel Snoop/213009-0000000e setting write format path: slin16 -> slin [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel Snoop/213009-0000000e setting read format path: slin -> slin16 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 From: ;tag=as3f810040 To: ;tag=as622102a0 Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f810040 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as622102a0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 (Checking To) --From tag as3f810040 --To-tag as622102a0 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:05] VERBOSE[14035] dial.c: Called zvonobot/79821116967 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[14039] chan_sip.c: Audio is at 17578 [Aug 18 10:34:05] VERBOSE[14039] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] VERBOSE[14039] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] VERBOSE[14039] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Initializing initreq for method INVITE - callid 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116964@178.62.121.41 SIP/2.0 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 3 [ 52]: From: ;tag=as22570a36 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 6 [ 60]: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] VERBOSE[14039] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #63 [Aug 18 10:34:05] DEBUG[14039] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (2) BYE - 8 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[13973] channel.c: Channel 0x7f0ca00ed5f0 'SIP/zvonobot-0000006f' allocated [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13973] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 setting write format path: slin16 -> slin16 [Aug 18 10:34:05] DEBUG[13973] res_stasis.c: calls_0: Subscribing to 213079 [Aug 18 10:34:05] DEBUG[13973] stasis/app.c: Channel '213079' is 1 interested in calls_0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 Max-Forwards: 70 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6be15af9 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Outgoing Call for 79821116961 [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Finding handler for bridges/c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:34:05] DEBUG[14005] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:05] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:05] DEBUG[13973] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13973] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0453a0d2 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6abcd5d1", response="23a984ce3130e223dae22cec8c1c0802" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 (Checking From) --From tag as6be15af9 --To-tag as0453a0d2 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:05] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Received bye, no owner, selfdestruct soon. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK43da7418;received=178.62.121.41 From: ;tag=as6be15af9 To: ;tag=as0453a0d2 Call-ID: 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 From: ;tag=as2618d3a5 To: ;tag=as6e1d4afe Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2618d3a5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e1d4afe [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 (Checking To) --From tag as2618d3a5 --To-tag as6e1d4afe [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 From: ;tag=as3edf3f1c To: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c1708ff;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3edf3f1c [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060 (Checking To) --From tag as3edf3f1c --To-tag [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '117823df7d1dc3622a802d2d2f745d6d@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:05] DEBUG[14005] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[14044] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[13993] stasis/app.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' is 4 interested in calls_0 [Aug 18 10:34:05] DEBUG[14045] http.c: HTTP opening session. Top level [Aug 18 10:34:05] DEBUG[14044] http.c: HTTP Request URI is /ari/channels/212992 [Aug 18 10:34:05] DEBUG[14045] http.c: HTTP Request URI is /ari/bridges/28c87384-44a9-4ebc-9328-4118df068e33/addChannel?channel=1629282842.223%2Crobot_213011 [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:05] DEBUG[14045] http.c: match request [ari/bridges/28c87384-44a9-4ebc-9328-4118df068e33/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:05] DEBUG[14038] channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 setting write format path: slin -> slin16 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP ooh, format changed from none to slin16 [Aug 18 10:34:05] DEBUG[14044] http.c: match request [ari/channels/212992] with handler [httpstatus] len 10 [Aug 18 10:34:05] DEBUG[14045] http.c: match request [ari/bridges/28c87384-44a9-4ebc-9328-4118df068e33/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[14045] http.c: match request [ari/bridges/28c87384-44a9-4ebc-9328-4118df068e33/addChannel] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[14045] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Finding handler for bridges/28c87384-44a9-4ebc-9328-4118df068e33/addChannel [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Finding handler for bridges [Aug 18 10:34:05] DEBUG[14044] http.c: match request [ari/channels/212992] with handler [phoneprov] len 9 [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:05] VERBOSE[14039] dial.c: Called zvonobot/79821116964 [Aug 18 10:34:05] DEBUG[14044] http.c: match request [ari/channels/212992] with handler [ari] len 3 [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:05] DEBUG[14044] http.c: Match made with [ari] [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Finding handler for channels/212992 [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Finding handler for channels [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Finding handler for 28c87384-44a9-4ebc-9328-4118df068e33 [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:05] DEBUG[14045] res_ari.c: No explicit handler found for 28c87384-44a9-4ebc-9328-4118df068e33. Using wildcard bridgeId. [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Finding handler for addChannel [Aug 18 10:34:05] DEBUG[14045] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:05] DEBUG[14045] stasis/control.c: 1629282842.223: Sending channel add_to_bridge command [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Finding handler for 212992 [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking channels create: Didn't match 212992 [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:05] DEBUG[14044] res_ari.c: Checking channels externalMedia: Didn't match 212992 [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Finding handler for bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:05] DEBUG[14044] res_ari.c: No explicit handler found for 212992. Using wildcard channelId. [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Finding handler for c66c6480-4085-4bd9-87d2-ee6f5748dcc3 [Aug 18 10:34:05] DEBUG[14036] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:05] DEBUG[14036] res_ari.c: No explicit handler found for c66c6480-4085-4bd9-87d2-ee6f5748dcc3. Using wildcard bridgeId. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 From: ;tag=as6d7500c6 To: ;tag=as14883ee4 Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6d7500c6 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as14883ee4 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 (Checking To) --From tag as6d7500c6 --To-tag as14883ee4 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (1) INVITE - 5 [Aug 18 10:34:05] DEBUG[14036] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: telling all channels to leave the party [Aug 18 10:34:05] DEBUG[14036] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:05] DEBUG[14036] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: queueing action type:13 sub:1001 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: calling stasis bridge destructor [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: calling simple_bridge technology stop [Aug 18 10:34:05] DEBUG[20534] bridge.c: Bridge c66c6480-4085-4bd9-87d2-ee6f5748dcc3: calling simple_bridge technology destructor [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3': is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[20620] stasis/app.c: bridge 'c66c6480-4085-4bd9-87d2-ee6f5748dcc3' unsubscribed from calls_0 [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: cache:157/bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3, detail: [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'cache:157/bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3': 0x7f0c9c03a6c0 destroyed [Aug 18 10:34:05] DEBUG[20620] stasis.c: Destroying topic. name: bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3, detail: [Aug 18 10:34:05] DEBUG[20620] stasis.c: Topic 'bridge:c66c6480-4085-4bd9-87d2-ee6f5748dcc3': 0x7f0c9c036820 destroyed [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[14041] chan_sip.c: Audio is at 17664 [Aug 18 10:34:05] DEBUG[13896] bridge_roles.c: Roles did not exist on channel Snoop/213011-0000000f [Aug 18 10:34:05] DEBUG[13896] stasis/control.c: 1629282842.223: Adding to bridge 28c87384-44a9-4ebc-9328-4118df068e33 [Aug 18 10:34:05] DEBUG[13896] stasis/app.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' is 1 interested in calls_0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 From: ;tag=as3d872a68 To: ;tag=as578d3717 Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" Content-Length: 0 <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:05] DEBUG[14047] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c080f3ea0(Snoop/213011-0000000f) is joining [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as578d3717 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:05] VERBOSE[14041] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag as578d3717 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:05] VERBOSE[14041] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1307958254 1307958254 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14926 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3d26e887 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] VERBOSE[14041] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[14036] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:05] DEBUG[14036] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1307958254 1307958254 IN IP4 178.62.121.41 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14926 RTP/AVP 0 8 101 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 (Checking To) --From tag as7d114a77 --To-tag as3d26e887 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Stopping retransmission on '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Got SDP version 1307958254 and unique parts [root 1307958254 IN IP4 178.62.121.41] [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1307958254 1307958254 IN IP4 178.62.121.41... OK. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:05] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:05] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Initializing initreq for method INVITE - callid 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116961@178.62.121.41 SIP/2.0 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 3 [ 52]: From: ;tag=as10d8c0eb [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 6 [ 60]: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:05] VERBOSE[14041] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:34:05] DEBUG[14041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4008d90) ICE set role failed; no ice instance [Aug 18 10:34:05] DEBUG[14047] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: pushing 0x7f0c080f3ea0(Snoop/213011-0000000f) [Aug 18 10:34:05] VERBOSE[14041] dial.c: Called zvonobot/79821116961 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:05] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP setting address on RTP instance [Aug 18 10:34:05] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0cb400c820 -- Strict RTP learning after remote address set to: 178.62.121.41:14926 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14926 [Aug 18 10:34:05] VERBOSE[14047] bridge_channel.c: Channel Snoop/213011-0000000f joined 'simple_bridge' stasis-bridge <28c87384-44a9-4ebc-9328-4118df068e33> [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00296c8) from 0x7f0c147e2330 to 0x7f0cb4008f68 [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb006fae8) from 0x7f0c147e2330 to 0x7f0cb4008f68 [Aug 18 10:34:05] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb000a9f8) from 0x7f0c147e2330 to 0x7f0cb4008f68 [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP ignoring duplicate property [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:05] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000001 setting read format path: alaw -> alaw [Aug 18 10:34:05] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000001 setting write format path: alaw -> alaw [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4008d90) DTLS - ast_rtp_activate rtp=0x7f0cb400c820 - setup and perform DTLS' [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb400c820) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:05] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb400c820) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:05] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:05] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Strict routing enforced for session 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:05] DEBUG[14047] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' can not use native RTP bridge as two channels are required [Aug 18 10:34:05] DEBUG[14047] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:05] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:05] DEBUG[14047] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[14047] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:05] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:05] DEBUG[14047] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[14047] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33 is already using the new technology. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117075@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c527273 Max-Forwards: 70 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:05] DEBUG[14047] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c080f3ea0(Snoop/213011-0000000f) is joining simple_bridge technology [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] VERBOSE[12869] dial.c: SIP/zvonobot-00000001 answered [Aug 18 10:34:05] VERBOSE[12869] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000001 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:05] DEBUG[12869] stasis/app.c: Channel '212965' is 2 interested in calls_0 [Aug 18 10:34:05] VERBOSE[12869] res_rtp_asterisk.c: 0x7f0cb400c820 -- Strict RTP switching to RTP target address 178.62.121.41:14926 as source [Aug 18 10:34:05] DEBUG[12869] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:05] DEBUG[12869] channel.c: Channel SIP/zvonobot-00000001 setting read format path: ulaw -> alaw [Aug 18 10:34:05] DEBUG[12869] channel.c: Channel SIP/zvonobot-00000001 setting write format path: alaw -> ulaw [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:05] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (1) INVITE - 5 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:05] DEBUG[13972] channel.c: Channel 0x7f0c9803da40 'SIP/zvonobot-00000070' allocated [Aug 18 10:34:05] DEBUG[13896] stasis/app.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' is 2 interested in calls_0 [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13972] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] DEBUG[14045] stasis/control.c: robot_213011: Sending channel add_to_bridge command [Aug 18 10:34:05] DEBUG[13951] channel.c: Channel 0x7f0c8c050630 'SIP/zvonobot-00000071' allocated [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13951] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:05] DEBUG[20585] chan_sip.c: Session timer started: 64 - 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 1768000ms [Aug 18 10:34:05] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:05] DEBUG[13951] res_stasis.c: calls_0: Subscribing to 213074 [Aug 18 10:34:05] DEBUG[13951] stasis/app.c: Channel '213074' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[13972] res_stasis.c: calls_0: Subscribing to 213078 [Aug 18 10:34:05] DEBUG[14048] chan_sip.c: Outgoing Call for 79821116966 [Aug 18 10:34:05] DEBUG[14048] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[13972] stasis/app.c: Channel '213078' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[14048] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[14048] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[14048] chan_sip.c: Audio is at 10288 [Aug 18 10:34:05] VERBOSE[14048] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] DEBUG[13972] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] VERBOSE[14048] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] VERBOSE[14048] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[14048] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[13972] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[13951] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:05] DEBUG[13951] http.c: HTTP closing session. Top level [Aug 18 10:34:05] DEBUG[14050] chan_sip.c: Outgoing Call for 79821116962 [Aug 18 10:34:05] DEBUG[14050] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:05] DEBUG[14050] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:05] DEBUG[14050] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:05] VERBOSE[14050] chan_sip.c: Audio is at 10498 [Aug 18 10:34:05] VERBOSE[14050] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:05] VERBOSE[14050] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:05] DEBUG[13961] channel.c: Channel 0x7f0ca80e0110 'SIP/zvonobot-00000072' allocated [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:05] DEBUG[13961] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:05] VERBOSE[14050] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:05] DEBUG[14050] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:05] DEBUG[13961] res_stasis.c: calls_0: Subscribing to 213075 [Aug 18 10:34:05] DEBUG[13961] stasis/app.c: Channel '213075' is 1 interested in calls_0 [Aug 18 10:34:05] DEBUG[14051] chan_sip.c: Outgoing Call for 79821116965 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:06] DEBUG[13961] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:06] DEBUG[13961] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:06] VERBOSE[14051] chan_sip.c: Audio is at 16396 [Aug 18 10:34:06] VERBOSE[14051] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:06] VERBOSE[14051] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:06] DEBUG[14054] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14054] http.c: HTTP Request URI is /ari/channels/213084?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116956&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14054] http.c: match request [ari/channels/213084] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14054] http.c: match request [ari/channels/213084] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14054] http.c: match request [ari/channels/213084] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14054] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14054] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Finding handler for channels/213084 [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Finding handler for 213084 [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking channels create: Didn't match 213084 [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14054] res_ari.c: Checking channels externalMedia: Didn't match 213084 [Aug 18 10:34:06] DEBUG[14054] res_ari.c: No explicit handler found for 213084. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14011] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 [Aug 18 10:34:06] DEBUG[14011] stasis/control.c: robot_213011: Adding to bridge 28c87384-44a9-4ebc-9328-4118df068e33 [Aug 18 10:34:06] DEBUG[14011] stasis/app.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' is 3 interested in calls_0 [Aug 18 10:34:06] DEBUG[14057] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) is joining [Aug 18 10:34:06] DEBUG[14059] http.c: HTTP opening session. Top level [Aug 18 10:34:06] VERBOSE[14051] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:06] DEBUG[14059] http.c: HTTP Request URI is /ari/channels/213086?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116954&callerId=74950493843 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 221776054 221776054 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10694 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:06] DEBUG[14059] http.c: match request [ari/channels/213086] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:06] DEBUG[14059] http.c: match request [ari/channels/213086] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Initializing initreq for method INVITE - callid 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116966@178.62.121.41 SIP/2.0 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 3 [ 52]: From: ;tag=as7eb98fd0 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 6 [ 60]: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] VERBOSE[14048] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #88 [Aug 18 10:34:06] DEBUG[14048] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Initializing initreq for method INVITE - callid 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:06] VERBOSE[14048] dial.c: Called zvonobot/79821116966 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116962@178.62.121.41 SIP/2.0 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 3 [ 52]: From: ;tag=as0a05f417 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 6 [ 60]: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:05 GMT [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] VERBOSE[14050] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #60 [Aug 18 10:34:06] DEBUG[14050] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14057] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: pushing 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) [Aug 18 10:34:06] VERBOSE[14050] dial.c: Called zvonobot/79821116962 [Aug 18 10:34:06] DEBUG[14059] http.c: match request [ari/channels/213086] with handler [ari] len 3 [Aug 18 10:34:06] VERBOSE[14057] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 joined 'simple_bridge' stasis-bridge <28c87384-44a9-4ebc-9328-4118df068e33> [Aug 18 10:34:06] DEBUG[14059] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 - start 1629282845.406866 answer 1629282845.560326 end 1629282846.075329 dur 0.668 bill 0.515 dispo ANSWERED [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:06] DEBUG[14063] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14059] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 [Aug 18 10:34:06] DEBUG[14057] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33'. Checking compatability for channels 'Snoop/213011-0000000f' and 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30' [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:06] DEBUG[14057] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' can not use native RTP bridge as could not get details [Aug 18 10:34:06] DEBUG[14057] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Initializing initreq for method INVITE - callid 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Finding handler for channels/213086 [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14057] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14057] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14057] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] DEBUG[14063] http.c: HTTP Request URI is /ari/channels/213085?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116955&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14057] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33 is already using the new technology. [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116965@178.62.121.41 SIP/2.0 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Finding handler for 213086 [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking channels create: Didn't match 213086 [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14059] res_ari.c: Checking channels externalMedia: Didn't match 213086 [Aug 18 10:34:06] DEBUG[14059] res_ari.c: No explicit handler found for 213086. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14063] http.c: match request [ari/channels/213085] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14069] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14063] http.c: match request [ari/channels/213085] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14057] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) is joining simple_bridge technology [Aug 18 10:34:06] DEBUG[14069] http.c: HTTP Request URI is /ari/channels/213087?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116953&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 setting read format path: slin16 -> slin16 [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14073] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14069] http.c: match request [ari/channels/213087] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel Snoop/213011-0000000f setting write format path: slin16 -> slin [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel Snoop/213011-0000000f setting read format path: slin -> slin16 [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 setting write format path: slin16 -> slin16 [Aug 18 10:34:06] DEBUG[14063] http.c: match request [ari/channels/213085] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14063] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14063] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Finding handler for channels/213085 [Aug 18 10:34:06] DEBUG[14069] http.c: match request [ari/channels/213087] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Finding handler for 213085 [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking channels create: Didn't match 213085 [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14063] res_ari.c: Checking channels externalMedia: Didn't match 213085 [Aug 18 10:34:06] DEBUG[14063] res_ari.c: No explicit handler found for 213085. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14069] http.c: match request [ari/channels/213087] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d218141 [Aug 18 10:34:06] DEBUG[14069] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14069] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 3 [ 52]: From: ;tag=as44c869a1 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Finding handler for channels/213087 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14011] stasis/app.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' is 4 interested in calls_0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6bcc6b47 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14045] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14074] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14045] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:06] DEBUG[14074] http.c: HTTP Request URI is /ari/channels/213089?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116951&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14073] http.c: HTTP Request URI is /ari/channels/213088?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116952&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: Allocating new SIP dialog for 435912cc394413a305590c711e393479@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14054] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9408bf50' [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) RTP allocated port 10010 [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE creating session 0.0.0.0:10010 (10010) [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE create [Aug 18 10:34:06] DEBUG[14073] http.c: match request [ari/channels/213088] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:06] DEBUG[14057] channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 setting write format path: slin -> slin16 [Aug 18 10:34:06] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP ooh, format changed from none to slin16 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE add system candidates [Aug 18 10:34:06] DEBUG[14081] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14081] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:06] DEBUG[14081] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14081] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14081] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14081] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14054] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14054] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE add candidate: 159.65.48.104:10010, 2130706431 [Aug 18 10:34:06] DEBUG[14054] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14054] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE add candidate: 10.131.0.10:10010, 2130706431 [Aug 18 10:34:06] DEBUG[14054] rtp_engine.c: RTP instance '0x7f0c9408bf50' is setup and ready to go [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE stopped [Aug 18 10:34:06] DEBUG[14054] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14054] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14054] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14054] res_rtp_asterisk.c: (0x7f0c9408bf50) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14054] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14054] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14054] chan_sip.c: SIP call-id changed from '435912cc394413a305590c711e393479@127.0.1.1:5060' to '14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14054] stasis.c: Creating topic. name: channel:213084, detail: [Aug 18 10:34:06] DEBUG[14054] stasis.c: Topic 'channel:213084': 0x7f0c940b36e0 created [Aug 18 10:34:06] DEBUG[14054] stasis.c: Creating topic. name: cache:295/channel:213084, detail: [Aug 18 10:34:06] DEBUG[14054] stasis.c: Topic 'cache:295/channel:213084': 0x7f0c940b4160 created [Aug 18 10:34:06] DEBUG[14074] http.c: match request [ari/channels/213089] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14074] http.c: match request [ari/channels/213089] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14074] http.c: match request [ari/channels/213089] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14074] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14074] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Finding handler for channels/213089 [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Finding handler for 213089 [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking channels create: Didn't match 213089 [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14074] res_ari.c: Checking channels externalMedia: Didn't match 213089 [Aug 18 10:34:06] DEBUG[14074] res_ari.c: No explicit handler found for 213089. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14073] http.c: match request [ari/channels/213088] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 221776054 221776054 IN IP4 178.62.121.41 [Aug 18 10:34:06] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10694 RTP/AVP 0 8 101 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:06] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14073] http.c: match request [ari/channels/213088] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 672, ms is 62 [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:06] DEBUG[14073] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14073] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Finding handler for 213087 [Aug 18 10:34:06] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking channels create: Didn't match 213087 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:06] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 752, ms is 114 [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: = Looking for Call ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 (Checking To) --From tag as2d218141 --To-tag as6bcc6b47 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Stopping retransmission on '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Got SDP version 221776054 and unique parts [root 221776054 IN IP4 178.62.121.41] [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 221776054 221776054 IN IP4 178.62.121.41... OK. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24032c40) ICE set role failed; no ice instance [Aug 18 10:34:06] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP setting address on RTP instance [Aug 18 10:34:06] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c2403c460 -- Strict RTP learning after remote address set to: 178.62.121.41:10694 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10694 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0130d88) from 0x7f0c147e2330 to 0x7f0c24032e18 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0130d08) from 0x7f0c147e2330 to 0x7f0c24032e18 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb007e538) from 0x7f0c147e2330 to 0x7f0c24032e18 [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP ignoring duplicate property [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:06] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000033 setting read format path: alaw -> alaw [Aug 18 10:34:06] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000033 setting write format path: alaw -> alaw [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24032c40) DTLS - ast_rtp_activate rtp=0x7f0c2403c460 - setup and perform DTLS' [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2403c460) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2403c460) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:06] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Strict routing enforced for session 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117025@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c50d44b Max-Forwards: 70 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Finding handler for channels/213088 [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 6 [ 60]: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:06 GMT [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] VERBOSE[14051] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 [Aug 18 10:34:06] DEBUG[14051] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: Checking channels externalMedia: Didn't match 213087 [Aug 18 10:34:06] DEBUG[14069] res_ari.c: No explicit handler found for 213087. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[13263] dial.c: SIP/zvonobot-00000033 answered [Aug 18 10:34:06] VERBOSE[13263] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000033 [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14084] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116989@178.62.121.41", nonce="6309d33d", response="47eab950ba6e0f9e0f2d586844a5a0a5" Date: Wed, 18 Aug 2021 10:34:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 558516628 558516629 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15904 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (1) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: Allocating new SIP dialog for 7d2aeea3167891247d3d506c09eaae42@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14063] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c093a50' [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) RTP allocated port 14468 [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE creating session 0.0.0.0:14468 (14468) [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE create [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 10 instead [Aug 18 10:34:06] DEBUG[14078] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Finding handler for 213088 [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking channels create: Didn't match 213088 [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14073] res_ari.c: Checking channels externalMedia: Didn't match 213088 [Aug 18 10:34:06] DEBUG[14073] res_ari.c: No explicit handler found for 213088. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[14051] dial.c: Called zvonobot/79821116965 [Aug 18 10:34:06] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (1) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (3) BYE - 8 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (1) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (4) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Session timer started: 102 - 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 1768000ms [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE add system candidates [Aug 18 10:34:06] DEBUG[14063] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14063] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE add candidate: 159.65.48.104:14468, 2130706431 [Aug 18 10:34:06] DEBUG[14063] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14063] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE add candidate: 10.131.0.10:14468, 2130706431 [Aug 18 10:34:06] DEBUG[14063] rtp_engine.c: RTP instance '0x7f0c9c093a50' is setup and ready to go [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE stopped [Aug 18 10:34:06] DEBUG[14063] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14063] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14063] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14063] res_rtp_asterisk.c: (0x7f0c9c093a50) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14063] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14063] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14063] chan_sip.c: SIP call-id changed from '7d2aeea3167891247d3d506c09eaae42@127.0.1.1:5060' to '12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:06] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:06] DEBUG[14078] http.c: HTTP Request URI is /ari/channels/213091?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116949&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 976, ms is 81 [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14078] http.c: match request [ari/channels/213091] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14078] http.c: match request [ari/channels/213091] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13263] stasis/app.c: Channel '213015' is 2 interested in calls_0 [Aug 18 10:34:06] VERBOSE[13263] res_rtp_asterisk.c: 0x7f0c2403c460 -- Strict RTP switching to RTP target address 178.62.121.41:10694 as source [Aug 18 10:34:06] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[14063] stasis.c: Creating topic. name: channel:213085, detail: [Aug 18 10:34:06] DEBUG[14063] stasis.c: Topic 'channel:213085': 0x7f0c9c0a1d40 created [Aug 18 10:34:06] DEBUG[14063] stasis.c: Creating topic. name: cache:296/channel:213085, detail: [Aug 18 10:34:06] DEBUG[14063] stasis.c: Topic 'cache:296/channel:213085': 0x7f0c9c0a27c0 created [Aug 18 10:34:06] DEBUG[13263] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:06] DEBUG[13263] channel.c: Channel SIP/zvonobot-00000033 setting read format path: ulaw -> alaw [Aug 18 10:34:06] DEBUG[13263] channel.c: Channel SIP/zvonobot-00000033 setting write format path: alaw -> ulaw [Aug 18 10:34:06] DEBUG[14084] http.c: HTTP Request URI is /ari/channels/213093?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116947&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14085] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14084] http.c: match request [ari/channels/213093] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14084] http.c: match request [ari/channels/213093] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14084] http.c: match request [ari/channels/213093] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14084] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14084] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Finding handler for channels/213093 [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Finding handler for 213093 [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking channels create: Didn't match 213093 [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14084] res_ari.c: Checking channels externalMedia: Didn't match 213093 [Aug 18 10:34:06] DEBUG[14084] res_ari.c: No explicit handler found for 213093. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14085] http.c: HTTP Request URI is /ari/channels/213090?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116950&callerId=74950493843 [Aug 18 10:34:06] DEBUG[14085] http.c: match request [ari/channels/213090] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14085] http.c: match request [ari/channels/213090] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14085] http.c: match request [ari/channels/213090] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14085] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14085] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Finding handler for channels/213090 [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[13975] channel.c: Channel 0x7f0ca4040e00 'SIP/zvonobot-00000073' allocated [Aug 18 10:34:06] DEBUG[14078] http.c: match request [ari/channels/213091] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:06] DEBUG[13975] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:06] DEBUG[14081] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:06] DEBUG[14087] http.c: HTTP Request URI is /ari/channels/213092?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116948&callerId=74950493843 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3725d74e [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:06] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:06] DEBUG[14078] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14087] http.c: match request [ari/channels/213092] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:06] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13975] res_stasis.c: calls_0: Subscribing to 213082 [Aug 18 10:34:06] DEBUG[14087] http.c: match request [ari/channels/213092] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14087] http.c: match request [ari/channels/213092] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14087] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as3725d74e [Aug 18 10:34:06] DEBUG[14087] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[13975] stasis/app.c: Channel '213082' is 1 interested in calls_0 [Aug 18 10:34:06] DEBUG[14089] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #51 [Aug 18 10:34:06] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 880, ms is 130 [Aug 18 10:34:06] DEBUG[14081] stasis.c: Creating topic. name: bridge:e8b160c4-f3ae-46ad-bda0-ffa245693ffa, detail: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Stopping retransmission on '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' of Request 104: Match Found [Aug 18 10:34:06] DEBUG[13958] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116987@178.62.121.41", nonce="38b12677", response="570676be99854ff0a2a1ea55211e3519" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937946687 937946688 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14624 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (4) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (1) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: Allocating new SIP dialog for 4ead21fa33a6e9c0420af501580e0ee6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14078] http.c: HTTP consuming request body [Aug 18 10:34:06] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:06] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:06] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:06] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 816, ms is 71 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 From: ;tag=as28933467 To: ;tag=as5b70cf89 Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1807314912 1807314912 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28933467 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5b70cf89 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1807314912 1807314912 IN IP4 178.62.121.41 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14088 RTP/AVP 0 8 101 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking To) --From tag as28933467 --To-tag as5b70cf89 [Aug 18 10:34:06] DEBUG[13975] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Outgoing Call for 79821116958 [Aug 18 10:34:06] DEBUG[13975] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Finding handler for 213090 [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking channels create: Didn't match 213090 [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14085] res_ari.c: Checking channels externalMedia: Didn't match 213090 [Aug 18 10:34:06] DEBUG[14085] res_ari.c: No explicit handler found for 213090. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Finding handler for channels/213092 [Aug 18 10:34:06] DEBUG[14089] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:06] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13958] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: Allocating new SIP dialog for 48ccf5643135bff1027740fa68f802a8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14059] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca8078fc0' [Aug 18 10:34:06] DEBUG[14081] stasis.c: Topic 'bridge:e8b160c4-f3ae-46ad-bda0-ffa245693ffa': 0x7f0cac07adb0 created [Aug 18 10:34:06] DEBUG[14081] stasis.c: Creating topic. name: cache:297/bridge:e8b160c4-f3ae-46ad-bda0-ffa245693ffa, detail: [Aug 18 10:34:06] DEBUG[14081] stasis.c: Topic 'cache:297/bridge:e8b160c4-f3ae-46ad-bda0-ffa245693ffa': 0x7f0cac098590 created [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Finding handler for channels/213091 [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Finding handler for channels [Aug 18 10:34:06] DEBUG[14089] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13884] stasis.c: Creating topic. name: channel:1629282846.255, detail: [Aug 18 10:34:06] DEBUG[13884] stasis.c: Topic 'channel:1629282846.255': 0x7f0ca406c5c0 created [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Finding handler for 213092 [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking channels create: Didn't match 213092 [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14087] res_ari.c: Checking channels externalMedia: Didn't match 213092 [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Got SDP version 1807314912 and unique parts [root 1807314912 IN IP4 178.62.121.41] [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1807314912 1807314912 IN IP4 178.62.121.41... OK. [Aug 18 10:34:06] DEBUG[14081] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' can not use native RTP bridge as two channels are required [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) RTP allocated port 10836 [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE creating session 0.0.0.0:10836 (10836) [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE create [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE add system candidates [Aug 18 10:34:06] DEBUG[14059] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14059] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE add candidate: 159.65.48.104:10836, 2130706431 [Aug 18 10:34:06] DEBUG[14059] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14059] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14087] res_ari.c: No explicit handler found for 213092. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14081] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[14081] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14081] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:06] DEBUG[14081] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] DEBUG[14091] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[13884] stasis.c: Creating topic. name: cache:298/channel:1629282846.255, detail: [Aug 18 10:34:06] DEBUG[13884] stasis.c: Topic 'cache:298/channel:1629282846.255': 0x7f0ca4043b10 created [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:06] DEBUG[14073] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca00fa8a0' [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) RTP allocated port 15512 [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE creating session 0.0.0.0:15512 (15512) [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE create [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE add system candidates [Aug 18 10:34:06] DEBUG[14073] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14073] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE add candidate: 159.65.48.104:15512, 2130706431 [Aug 18 10:34:06] DEBUG[14073] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14073] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE add candidate: 10.131.0.10:15512, 2130706431 [Aug 18 10:34:06] DEBUG[14073] rtp_engine.c: RTP instance '0x7f0ca00fa8a0' is setup and ready to go [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE stopped [Aug 18 10:34:06] DEBUG[14073] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14073] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14073] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14073] res_rtp_asterisk.c: (0x7f0ca00fa8a0) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14073] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14089] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Finding handler for 213091 [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking channels create: Didn't match 213091 [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14078] res_ari.c: Checking channels externalMedia: Didn't match 213091 [Aug 18 10:34:06] DEBUG[14078] res_ari.c: No explicit handler found for 213091. Using wildcard channelId. [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14089] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14081] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology constructor [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE add candidate: 10.131.0.10:10836, 2130706431 [Aug 18 10:34:06] DEBUG[14081] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology start [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14089] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14059] rtp_engine.c: RTP instance '0x7f0ca8078fc0' is setup and ready to go [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE set role failed; no ice instance [Aug 18 10:34:06] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTCP setting address on RTP instance [Aug 18 10:34:06] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c2400e650 -- Strict RTP learning after remote address set to: 178.62.121.41:14088 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14088 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00ccfd8) from 0x7f0c147e2330 to 0x7f0c2400b9a8 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0129f88) from 0x7f0c147e2330 to 0x7f0c2400b9a8 [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb002d948) from 0x7f0c147e2330 to 0x7f0c2400b9a8 [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTCP ignoring duplicate property [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:06] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000004 setting read format path: alaw -> alaw [Aug 18 10:34:06] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000004 setting write format path: alaw -> alaw [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS - ast_rtp_activate rtp=0x7f0c2400e650 - setup and perform DTLS' [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400e650) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400e650) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:06] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:06] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:06] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117073@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20268c04 Max-Forwards: 70 From: ;tag=as28933467 To: ;tag=as5b70cf89 Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 832, ms is 72 [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[12888] dial.c: SIP/zvonobot-00000004 answered [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14081] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 848, ms is 73 [Aug 18 10:34:06] DEBUG[14081] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[14092] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 832, ms is 72 [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: Allocating new SIP dialog for 4f613a971518949500a74872732960ae@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14069] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c980b5ec0' [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) RTP allocated port 18262 [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE creating session 0.0.0.0:18262 (18262) [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE create [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE add system candidates [Aug 18 10:34:06] DEBUG[14069] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14069] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE add candidate: 159.65.48.104:18262, 2130706431 [Aug 18 10:34:06] DEBUG[14092] http.c: HTTP Request URI is /ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/addChannel?channel=212965 [Aug 18 10:34:06] DEBUG[14073] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 640, ms is 60 [Aug 18 10:34:06] DEBUG[14092] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[14091] http.c: HTTP Request URI is /ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[13947] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP audio difference is 1024, ms is 84 [Aug 18 10:34:06] VERBOSE[12888] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000004 [Aug 18 10:34:06] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[14089] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[14089] stasis.c: Creating topic. name: bridge:3fc9ee09-2746-49ab-833c-6c9b37b1bb83, detail: [Aug 18 10:34:06] DEBUG[14089] stasis.c: Topic 'bridge:3fc9ee09-2746-49ab-833c-6c9b37b1bb83': 0x7f0c08062730 created [Aug 18 10:34:06] DEBUG[14089] stasis.c: Creating topic. name: cache:299/bridge:3fc9ee09-2746-49ab-833c-6c9b37b1bb83, detail: [Aug 18 10:34:06] DEBUG[14089] stasis.c: Topic 'cache:299/bridge:3fc9ee09-2746-49ab-833c-6c9b37b1bb83': 0x7f0c0806bc10 created [Aug 18 10:34:06] DEBUG[14089] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' can not use native RTP bridge as two channels are required [Aug 18 10:34:06] DEBUG[14089] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[14089] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14089] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:06] DEBUG[14089] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] DEBUG[14089] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling simple_bridge technology constructor [Aug 18 10:34:06] DEBUG[14089] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling simple_bridge technology start [Aug 18 10:34:06] DEBUG[12888] stasis/app.c: Channel '212967' is 2 interested in calls_0 [Aug 18 10:34:06] VERBOSE[12888] res_rtp_asterisk.c: 0x7f0c2400e650 -- Strict RTP switching to RTP target address 178.62.121.41:14088 as source [Aug 18 10:34:06] DEBUG[12888] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:06] DEBUG[12888] channel.c: Channel SIP/zvonobot-00000004 setting read format path: ulaw -> alaw [Aug 18 10:34:06] DEBUG[12888] channel.c: Channel SIP/zvonobot-00000004 setting write format path: alaw -> ulaw [Aug 18 10:34:06] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14092] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14089] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:06] VERBOSE[14090] chan_sip.c: Audio is at 11576 [Aug 18 10:34:06] VERBOSE[14090] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:06] VERBOSE[14090] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:06] VERBOSE[14090] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Initializing initreq for method INVITE - callid 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116958@178.62.121.41 SIP/2.0 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 3 [ 52]: From: ;tag=as15514e30 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 6 [ 60]: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:06 GMT [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:06] VERBOSE[14090] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #111 [Aug 18 10:34:06] DEBUG[14090] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14091] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE stopped [Aug 18 10:34:06] DEBUG[14059] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14059] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14059] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14059] res_rtp_asterisk.c: (0x7f0ca8078fc0) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14059] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14059] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14092] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/addChannel] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[14091] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 768, ms is 116 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14091] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[14093] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14093] http.c: HTTP Request URI is /ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/addChannel?channel=213015 [Aug 18 10:34:06] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116976@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8 Max-Forwards: 70 From: ;tag=as7a3cd0ea To: Contact: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1074146549 1074146549 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10716 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Destroying SIP dialog 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS stop [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS srtp - stopped timeout timer' [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) DTLS srtp - stopped timeout timer' [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3401c090) ICE RTP transport deallocating [Aug 18 10:34:06] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3401c090' [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (4) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116996@178.62.121.41", nonce="268549f1", response="70b2be9ce3d5657a7453cacd989d7b2d" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1795318273 1795318274 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10612 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116975@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787 Max-Forwards: 70 From: ;tag=as5a5dd50f To: Contact: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 998972 998972 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14089] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[14092] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[14094] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Finding handler for bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/addChannel [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14091] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Finding handler for e8b160c4-f3ae-46ad-bda0-ffa245693ffa [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14092] res_ari.c: No explicit handler found for e8b160c4-f3ae-46ad-bda0-ffa245693ffa. Using wildcard bridgeId. [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Finding handler for addChannel [Aug 18 10:34:06] DEBUG[14092] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:06] DEBUG[14092] stasis/control.c: 212965: Sending channel add_to_bridge command [Aug 18 10:34:06] DEBUG[14073] chan_sip.c: SIP call-id changed from '4ead21fa33a6e9c0420af501580e0ee6@127.0.1.1:5060' to '62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14073] stasis.c: Creating topic. name: channel:213088, detail: [Aug 18 10:34:06] DEBUG[14073] stasis.c: Topic 'channel:213088': 0x7f0ca00de400 created [Aug 18 10:34:06] DEBUG[14073] stasis.c: Creating topic. name: cache:300/channel:213088, detail: [Aug 18 10:34:06] DEBUG[14073] stasis.c: Topic 'cache:300/channel:213088': 0x7f0ca00ea4e0 created [Aug 18 10:34:06] DEBUG[14094] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14069] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] VERBOSE[14090] dial.c: Called zvonobot/79821116958 [Aug 18 10:34:06] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[12869] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000001 [Aug 18 10:34:06] DEBUG[12869] stasis/control.c: 212965: Adding to bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa [Aug 18 10:34:06] DEBUG[12869] stasis/app.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' is 1 interested in calls_0 [Aug 18 10:34:06] DEBUG[14094] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 752, ms is 67 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:06] DEBUG[14095] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is joining [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Finding handler for bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play [Aug 18 10:34:06] DEBUG[14094] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14094] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14069] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14094] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: Allocating new SIP dialog for 7782ccf55c39e3474dbf240f4eeb8023@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14074] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca4082ac0' [Aug 18 10:34:06] DEBUG[14059] chan_sip.c: SIP call-id changed from '48ccf5643135bff1027740fa68f802a8@127.0.1.1:5060' to '276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) RTP allocated port 18668 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #31 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #31)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116988@178.62.121.41", nonce="16d33e27", response="53fe5260b10bfea8d64eb009dc4c8466" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1378706391 1378706392 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14750 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14093] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/addChannel] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:06] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE creating session 0.0.0.0:18668 (18668) [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE create [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Session timer started: 84 - 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 1768000ms [Aug 18 10:34:06] DEBUG[14059] stasis.c: Creating topic. name: channel:213086, detail: [Aug 18 10:34:06] DEBUG[14059] stasis.c: Topic 'channel:213086': 0x7f0ca8077db0 created [Aug 18 10:34:06] DEBUG[14059] stasis.c: Creating topic. name: cache:301/channel:213086, detail: [Aug 18 10:34:06] DEBUG[14059] stasis.c: Topic 'cache:301/channel:213086': 0x7f0ca8005f90 created [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14095] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: pushing 0x2c6fb50(SIP/zvonobot-00000001) [Aug 18 10:34:06] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP got report of 100 bytes from 178.62.121.41:14675 [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE add system candidates [Aug 18 10:34:06] DEBUG[14074] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14074] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE add candidate: 159.65.48.104:18668, 2130706431 [Aug 18 10:34:06] DEBUG[14074] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14074] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE add candidate: 10.131.0.10:18668, 2130706431 [Aug 18 10:34:06] DEBUG[14074] rtp_engine.c: RTP instance '0x7f0ca4082ac0' is setup and ready to go [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) ICE stopped [Aug 18 10:34:06] DEBUG[14074] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14074] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14074] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14074] res_rtp_asterisk.c: (0x7f0ca4082ac0) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14074] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14074] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14074] chan_sip.c: SIP call-id changed from '7782ccf55c39e3474dbf240f4eeb8023@127.0.1.1:5060' to '01fa931654c30892638dda461f55f7f2@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14074] stasis.c: Creating topic. name: channel:213089, detail: [Aug 18 10:34:06] DEBUG[14074] stasis.c: Topic 'channel:213089': 0x7f0ca40594d0 created [Aug 18 10:34:06] DEBUG[14074] stasis.c: Creating topic. name: cache:302/channel:213089, detail: [Aug 18 10:34:06] DEBUG[14074] stasis.c: Topic 'cache:302/channel:213089': 0x7f0ca4059e70 created [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[14094] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] VERBOSE[14095] bridge_channel.c: Channel SIP/zvonobot-00000001 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:06] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: Allocating new SIP dialog for 5102b1612f972f0771c5c77202912bc2@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14087] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1012c700' [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) RTP allocated port 11608 [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE creating session 0.0.0.0:11608 (11608) [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE create [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE add system candidates [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[13870] stasis.c: Creating topic. name: channel:1629282846.257, detail: [Aug 18 10:34:06] DEBUG[13870] stasis.c: Topic 'channel:1629282846.257': 0x7f0c90010b00 created [Aug 18 10:34:06] DEBUG[13870] stasis.c: Creating topic. name: cache:303/channel:1629282846.257, detail: [Aug 18 10:34:06] DEBUG[13870] stasis.c: Topic 'cache:303/channel:1629282846.257': 0x7f0c9004ec10 created [Aug 18 10:34:06] DEBUG[14094] stasis.c: Creating topic. name: bridge:9cefb3ad-33ea-4a52-96a1-42b677d6802c, detail: [Aug 18 10:34:06] DEBUG[14087] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE add candidate: 10.131.0.10:18262, 2130706431 [Aug 18 10:34:06] DEBUG[14069] rtp_engine.c: RTP instance '0x7f0c980b5ec0' is setup and ready to go [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) ICE stopped [Aug 18 10:34:06] DEBUG[14069] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14069] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14069] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14069] res_rtp_asterisk.c: (0x7f0c980b5ec0) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14069] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14069] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14069] chan_sip.c: SIP call-id changed from '4f613a971518949500a74872732960ae@127.0.1.1:5060' to '5993fccb0e95740465a028667804b469@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14069] stasis.c: Creating topic. name: channel:213087, detail: [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 1152, ms is 92 [Aug 18 10:34:06] DEBUG[14093] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP got report of 100 bytes from 178.62.121.41:15419 [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Finding handler for bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/addChannel [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: Allocating new SIP dialog for 13cef1b76f123c0a366f367e4e47e74b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14085] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c42620' [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) RTP allocated port 17384 [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE creating session 0.0.0.0:17384 (17384) [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE create [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE add system candidates [Aug 18 10:34:06] DEBUG[14085] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14085] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE add candidate: 159.65.48.104:17384, 2130706431 [Aug 18 10:34:06] DEBUG[14085] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14085] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE add candidate: 10.131.0.10:17384, 2130706431 [Aug 18 10:34:06] DEBUG[14085] rtp_engine.c: RTP instance '0x2c42620' is setup and ready to go [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) ICE stopped [Aug 18 10:34:06] DEBUG[14085] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14085] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14085] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14085] res_rtp_asterisk.c: (0x2c42620) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14085] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14085] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14069] stasis.c: Topic 'channel:213087': 0x7f0c98082fe0 created [Aug 18 10:34:06] DEBUG[14069] stasis.c: Creating topic. name: cache:304/channel:213087, detail: [Aug 18 10:34:06] DEBUG[14069] stasis.c: Topic 'cache:304/channel:213087': 0x7f0c98083980 created [Aug 18 10:34:06] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Finding handler for aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE add candidate: 159.65.48.104:11608, 2130706431 [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14094] stasis.c: Topic 'bridge:9cefb3ad-33ea-4a52-96a1-42b677d6802c': 0x7f0c2c0920e0 created [Aug 18 10:34:06] WARNING[13151] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000002;1 [Aug 18 10:34:06] DEBUG[14091] res_ari.c: No explicit handler found for aba705f1-c39f-408a-8a02-8c7f66ee7c7d. Using wildcard bridgeId. [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Finding handler for play [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:06] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[14087] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP got report of 76 bytes from 178.62.121.41:18793 [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Finding handler for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50409 - state 0 (Unknown) [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:06] DEBUG[14091] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:06] DEBUG[14094] stasis.c: Creating topic. name: cache:305/bridge:9cefb3ad-33ea-4a52-96a1-42b677d6802c, detail: [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13879] channel.c: Channel 0x7f0c980b5300 'Announcer/ARI-00000027;2' allocated [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:06] DEBUG[13879] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:06] DEBUG[13879] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000027;1' [Aug 18 10:34:06] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 1088, ms is 88 [Aug 18 10:34:06] DEBUG[13153] channel.c: Channel 0x7f0c20014960 'Announcer/ARI-00000002;1' destroying [Aug 18 10:34:06] DEBUG[13891] res_rtp_asterisk.c: (0x7f0c7804a920) RTP audio difference is 848, ms is 73 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 From: ;tag=as6ceaa437 To: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:06] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50409' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Aug 18 10:34:06] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:06] DEBUG[13153] stasis.c: Destroying topic. name: cache:57/channel:1629282829.48, detail: [Aug 18 10:34:06] DEBUG[13153] stasis.c: Topic 'cache:57/channel:1629282829.48': 0x7f0c20033060 destroyed [Aug 18 10:34:06] DEBUG[13151] bridge_channel.c: Setting 0x7f0c2001ab20(Announcer/ARI-00000002;2) state from:0 to:1 [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13151] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pulling 0x7f0c2001ab20(Announcer/ARI-00000002;2) [Aug 18 10:34:06] VERBOSE[13151] bridge_channel.c: Channel Announcer/ARI-00000002;2 left 'softmix' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:34:06] DEBUG[13151] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0c2001ab20(Announcer/ARI-00000002;2) is leaving softmix technology [Aug 18 10:34:06] DEBUG[13151] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c'. Checking compatability for channels 'SIP/zvonobot-00000000' and 'Recorder/ARI-00000000;2' [Aug 18 10:34:06] DEBUG[13151] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as channel 'SIP/zvonobot-00000000' has features which prevent it [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[13151] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] VERBOSE[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: switching from softmix technology to simple_bridge [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology constructor [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0cac00a6f0(SIP/zvonobot-00000000) to dummy bridge temporarily [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: moving 0x7f0ca400f4f0(Recorder/ARI-00000000;2) to dummy bridge temporarily [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is leaving softmix technology (dummy) [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is leaving softmix technology (dummy) [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology stop [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is joining simple_bridge technology [Aug 18 10:34:06] DEBUG[13151] channel.c: Channel Recorder/ARI-00000000;2 setting read format path: slin -> slin [Aug 18 10:34:06] DEBUG[13151] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is joining simple_bridge technology [Aug 18 10:34:06] DEBUG[13151] channel.c: Channel Recorder/ARI-00000000;2 setting read format path: slin -> slin [Aug 18 10:34:06] DEBUG[13151] channel.c: Channel Recorder/ARI-00000000;2 setting write format path: slin -> slin [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology start [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: deferring softmix technology destructor [Aug 18 10:34:06] DEBUG[13151] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: queueing action type:13 sub:1000 [Aug 18 10:34:06] DEBUG[13151] channel.c: Channel 0x7f0c2001ad30 'Announcer/ARI-00000002;2' hanging up. Refs: 2 [Aug 18 10:34:06] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:06] DEBUG[20534] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: Waiting for mixing thread to die. [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 928, ms is 78 [Aug 18 10:34:06] DEBUG[13153] stasis.c: Destroying topic. name: channel:1629282829.48, detail: [Aug 18 10:34:06] DEBUG[13058] channel.c: Recorder/ARI-00000000;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:06] DEBUG[13152] bridge_softmix.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: stopping mixing thread [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13947] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP audio difference is 672, ms is 62 [Aug 18 10:34:06] DEBUG[13153] stasis.c: Topic 'channel:1629282829.48': 0x7f0c20010a40 destroyed [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE add candidate: 10.131.0.10:11608, 2130706431 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:06] DEBUG[14093] res_ari.c: No explicit handler found for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83. Using wildcard bridgeId. [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Finding handler for addChannel [Aug 18 10:34:06] DEBUG[14093] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:06] DEBUG[14087] rtp_engine.c: RTP instance '0x7f0c1012c700' is setup and ready to go [Aug 18 10:34:06] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 640, ms is 60 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47fa7625;received=159.65.48.104 [Aug 18 10:34:06] DEBUG[14099] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c980404a0(Announcer/ARI-00000027;2) is joining [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ceaa437 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13263] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000033 [Aug 18 10:34:06] DEBUG[14085] chan_sip.c: SIP call-id changed from '13cef1b76f123c0a366f367e4e47e74b@127.0.1.1:5060' to '41cffb51539db62640feb00322cb29ef@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[14085] stasis.c: Creating topic. name: channel:213090, detail: [Aug 18 10:34:06] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 960, ms is 80 [Aug 18 10:34:06] DEBUG[14094] stasis.c: Topic 'cache:305/bridge:9cefb3ad-33ea-4a52-96a1-42b677d6802c': 0x7f0c2c0c3220 created [Aug 18 10:34:06] DEBUG[14094] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:06] DEBUG[14094] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[14094] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14094] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:06] DEBUG[14094] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] DEBUG[14094] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: calling simple_bridge technology constructor [Aug 18 10:34:06] DEBUG[14094] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: calling simple_bridge technology start [Aug 18 10:34:06] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTP audio difference is 664, ms is 103 [Aug 18 10:34:06] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:06] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP audio difference is 776, ms is 117 [Aug 18 10:34:06] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14093] stasis/control.c: 213015: Sending channel add_to_bridge command [Aug 18 10:34:06] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: Allocating new SIP dialog for 7a32c88532cbf9200a206c5e410460de@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14084] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb404d3b0' [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) RTP allocated port 10588 [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE creating session 0.0.0.0:10588 (10588) [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE create [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE add system candidates [Aug 18 10:34:06] DEBUG[14084] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14084] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE add candidate: 159.65.48.104:10588, 2130706431 [Aug 18 10:34:06] DEBUG[14084] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[14084] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE add candidate: 10.131.0.10:10588, 2130706431 [Aug 18 10:34:06] DEBUG[14084] rtp_engine.c: RTP instance '0x7f0cb404d3b0' is setup and ready to go [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE stopped [Aug 18 10:34:06] DEBUG[14084] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14084] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14084] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14084] res_rtp_asterisk.c: (0x7f0cb404d3b0) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14084] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14084] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14084] chan_sip.c: SIP call-id changed from '7a32c88532cbf9200a206c5e410460de@127.0.1.1:5060' to '63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14084] stasis.c: Creating topic. name: channel:213093, detail: [Aug 18 10:34:06] DEBUG[14084] stasis.c: Topic 'channel:213093': 0x7f0cb406a650 created [Aug 18 10:34:06] DEBUG[14084] stasis.c: Creating topic. name: cache:306/channel:213093, detail: [Aug 18 10:34:06] DEBUG[14084] stasis.c: Topic 'cache:306/channel:213093': 0x7f0cb4045d70 created [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) ICE stopped [Aug 18 10:34:06] DEBUG[14087] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[13056] channel.c: SIP/zvonobot-00000000: Dropping redundant connected line update "" <>. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: Allocating new SIP dialog for 1295403064a660c724055eb209f633de@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:06] DEBUG[14078] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0134510' [Aug 18 10:34:06] DEBUG[14087] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14087] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[13208] res_rtp_asterisk.c: (0x7f0c1007bd40) RTP audio difference is 840, ms is 125 [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060 (Checking To) --From tag as6ceaa437 --To-tag [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a5721714973d8ea7cd3306353c96cf0@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:06] DEBUG[14099] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: pushing 0x7f0c980404a0(Announcer/ARI-00000027;2) [Aug 18 10:34:06] DEBUG[14094] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP audio difference is 792, ms is 119 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:06] DEBUG[14087] res_rtp_asterisk.c: (0x7f0c1012c700) RTCP setup on RTP instance [Aug 18 10:34:06] DEBUG[14094] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13680] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14085] stasis.c: Topic 'channel:213090': 0x2c64460 created [Aug 18 10:34:06] DEBUG[14085] stasis.c: Creating topic. name: cache:307/channel:213090, detail: [Aug 18 10:34:06] DEBUG[14085] stasis.c: Topic 'cache:307/channel:213090': 0x2c23580 created [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14107] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14095] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' can not use native RTP bridge as two channels are required [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) RTP allocated port 18068 [Aug 18 10:34:06] DEBUG[14095] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:06] DEBUG[14095] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14095] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:06] VERBOSE[14087] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14095] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:06] DEBUG[14095] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa is already using the new technology. [Aug 18 10:34:06] DEBUG[14095] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is joining simple_bridge technology [Aug 18 10:34:06] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 656, ms is 61 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (2) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[13982] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #111 (1) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #111)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[14107] http.c: HTTP Request URI is /ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/addChannel?channel=212967 [Aug 18 10:34:06] DEBUG[13947] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP audio difference is 832, ms is 72 [Aug 18 10:34:06] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14087] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14087] chan_sip.c: SIP call-id changed from '5102b1612f972f0771c5c77202912bc2@127.0.1.1:5060' to '3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14087] stasis.c: Creating topic. name: channel:213092, detail: [Aug 18 10:34:06] DEBUG[14087] stasis.c: Topic 'channel:213092': 0x7f0c101507e0 created [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE creating session 0.0.0.0:18068 (18068) [Aug 18 10:34:06] DEBUG[14107] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13978] channel.c: Channel 0x7f0cac095830 'SIP/zvonobot-00000074' allocated [Aug 18 10:34:06] DEBUG[14107] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE create [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:06] DEBUG[13978] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:06] DEBUG[14087] stasis.c: Creating topic. name: cache:308/channel:213092, detail: [Aug 18 10:34:06] DEBUG[14099] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:06] DEBUG[14087] stasis.c: Topic 'cache:308/channel:213092': 0x7f0c1004f3e0 created [Aug 18 10:34:06] DEBUG[14107] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/addChannel] with handler [ari] len 3 [Aug 18 10:34:06] VERBOSE[14099] bridge_channel.c: Channel Announcer/ARI-00000027;2 joined 'simple_bridge' stasis-bridge <0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3> [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (3) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:06] DEBUG[14107] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Finding handler for bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/addChannel [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE add system candidates [Aug 18 10:34:06] DEBUG[14078] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:06] DEBUG[14078] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (4) INVITE - 5 [Aug 18 10:34:06] DEBUG[12869] stasis/app.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' is 2 interested in calls_0 [Aug 18 10:34:06] DEBUG[14092] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:06] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:06] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14092] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[13263] stasis/control.c: 213015: Adding to bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 [Aug 18 10:34:06] DEBUG[13263] stasis/app.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' is 1 interested in calls_0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (5) INVITE - 5 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE add candidate: 159.65.48.104:18068, 2130706431 [Aug 18 10:34:06] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP changing ssrc from 1000195339 to 2119175922 due to a source change [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:06] DEBUG[14099] bridge.c: Chose bridge technology softmix [Aug 18 10:34:06] VERBOSE[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: switching from simple_bridge technology to softmix [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling softmix technology constructor [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) to dummy bridge temporarily [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c20086d10(Recorder/ARI-00000023;2) to dummy bridge temporarily [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is leaving simple_bridge technology (dummy) [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology stop [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c980404a0(Announcer/ARI-00000027;2) is joining softmix technology [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Announcer/ARI-00000027;2: [Aug 18 10:34:06] DEBUG[14099] channel.c: Channel Announcer/ARI-00000027;2 setting write format path: slin -> slin [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: [Aug 18 10:34:06] DEBUG[14078] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:06] DEBUG[20580] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[20580] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Finding handler for 9cefb3ad-33ea-4a52-96a1-42b677d6802c [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14107] res_ari.c: No explicit handler found for 9cefb3ad-33ea-4a52-96a1-42b677d6802c. Using wildcard bridgeId. [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Finding handler for addChannel [Aug 18 10:34:06] DEBUG[14107] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:06] DEBUG[14107] stasis/control.c: 212967: Sending channel add_to_bridge command [Aug 18 10:34:06] DEBUG[14078] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE add candidate: 10.131.0.10:18068, 2130706431 [Aug 18 10:34:06] DEBUG[14078] rtp_engine.c: RTP instance '0x7f0cb0134510' is setup and ready to go [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) ICE stopped [Aug 18 10:34:06] DEBUG[14078] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:06] DEBUG[14078] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:06] DEBUG[14078] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:06] DEBUG[14078] res_rtp_asterisk.c: (0x7f0cb0134510) RTCP setup on RTP instance [Aug 18 10:34:06] VERBOSE[14078] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:06] DEBUG[14078] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 From: ;tag=as366f0ed0 To: ;tag=as0a7e373f Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a7e373f [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: = Looking for Call ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 (Checking To) --From tag as366f0ed0 --To-tag as0a7e373f [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Announcer/ARI-00000027;2: [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Announcer/ARI-00000027;2: Not in SFU mode [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is joining softmix technology [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: SIP/zvonobot-0000002f: [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: SIP/zvonobot-0000002f: [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: SIP/zvonobot-0000002f: Not in SFU mode [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is joining softmix technology [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Recorder/ARI-00000023;2: [Aug 18 10:34:06] DEBUG[14099] channel.c: Channel Recorder/ARI-00000023;2 setting write format path: slin -> slin [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:06] DEBUG[14099] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Recorder/ARI-00000023;2: [Aug 18 10:34:06] DEBUG[14099] bridge_softmix.c: Recorder/ARI-00000023;2: Not in SFU mode [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling softmix technology start [Aug 18 10:34:06] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology destructor [Aug 18 10:34:06] DEBUG[14111] http.c: HTTP opening session. Top level [Aug 18 10:34:06] DEBUG[14111] http.c: HTTP Request URI is /ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/record?name=212965_MdQEAXTNGgfRSCaEibwitrXymNroStbk&format=wav [Aug 18 10:34:06] DEBUG[14111] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/record] with handler [httpstatus] len 10 [Aug 18 10:34:06] DEBUG[13879] res_stasis_playback.c: 1629282843.231: Sending play(sound:silence/2) command [Aug 18 10:34:06] DEBUG[14113] bridge_softmix.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: starting mixing thread [Aug 18 10:34:06] DEBUG[14078] chan_sip.c: SIP call-id changed from '1295403064a660c724055eb209f633de@127.0.1.1:5060' to '1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060' [Aug 18 10:34:06] DEBUG[14111] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/record] with handler [phoneprov] len 9 [Aug 18 10:34:06] DEBUG[13879] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:06] DEBUG[14111] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/record] with handler [ari] len 3 [Aug 18 10:34:06] DEBUG[14111] http.c: Match made with [ari] [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Finding handler for bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Finding handler for bridges [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Stopping retransmission on '636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Finding handler for e8b160c4-f3ae-46ad-bda0-ffa245693ffa [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:06] DEBUG[14111] res_ari.c: No explicit handler found for e8b160c4-f3ae-46ad-bda0-ffa245693ffa. Using wildcard bridgeId. [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Finding handler for record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:06] DEBUG[14111] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:06] DEBUG[14111] stasis.c: Creating topic. name: channel:1629282846.265, detail: [Aug 18 10:34:06] DEBUG[14111] stasis.c: Topic 'channel:1629282846.265': 0x7f0c3c08e4f0 created [Aug 18 10:34:06] DEBUG[14111] stasis.c: Creating topic. name: cache:309/channel:1629282846.265, detail: [Aug 18 10:34:06] DEBUG[14111] stasis.c: Topic 'cache:309/channel:1629282846.265': 0x7f0c3c08ef20 created [Aug 18 10:34:06] DEBUG[13978] res_stasis.c: calls_0: Subscribing to 213081 [Aug 18 10:34:06] DEBUG[13879] http.c: HTTP closing session. Top level [Aug 18 10:34:06] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:06] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:06] DEBUG[14112] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c2c006860(SIP/zvonobot-00000033) is joining [Aug 18 10:34:06] DEBUG[14078] stasis.c: Creating topic. name: channel:213091, detail: [Aug 18 10:34:06] DEBUG[12888] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000004 [Aug 18 10:34:06] DEBUG[12888] stasis/control.c: 212967: Adding to bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c [Aug 18 10:34:06] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117020@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b Max-Forwards: 70 From: ;tag=as366f0ed0 To: ;tag=as0a7e373f Contact: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:06] DEBUG[14078] stasis.c: Topic 'channel:213091': 0x7f0cb011d800 created [Aug 18 10:34:06] DEBUG[14078] stasis.c: Creating topic. name: cache:310/channel:213091, detail: [Aug 18 10:34:06] DEBUG[14078] stasis.c: Topic 'cache:310/channel:213091': 0x7f0cb011d9c0 created [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:06] DEBUG[12888] stasis/app.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' is 1 interested in calls_0 [Aug 18 10:34:06] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:07] VERBOSE[13280] dial.c: SIP/zvonobot-00000036 is busy [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6109ms with no response [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Hanging up call 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13280] channel.c: Channel 0x7f0c34047590 'SIP/zvonobot-00000036' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5af3618d5953b3bf51714f3426a1ecf6@159.65.48.104:5060' Method: BYE [Aug 18 10:34:07] DEBUG[13767] channel.c: Channel 0x7f0c2c0a7290 'SIP/zvonobot-0000005a' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[14116] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c20083a40(SIP/zvonobot-00000004) is joining [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000036 - start 1629282832.235549 answer 0.000000 end 1629282847.006099 dur 14.770 bill 1629282847.006 dispo BUSY [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS stop [Aug 18 10:34:07] DEBUG[14112] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: pushing 0x7f0c2c006860(SIP/zvonobot-00000033) [Aug 18 10:34:07] VERBOSE[14112] bridge_channel.c: Channel SIP/zvonobot-00000033 joined 'simple_bridge' stasis-bridge <3fc9ee09-2746-49ab-833c-6c9b37b1bb83> [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70012180) ICE RTP transport deallocating [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c70012180' [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #111 (2) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #111)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (3) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (4) BYE - 8 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14112] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[13695] audiohook.c: Audiohook 0x7f0cb011ab60 has stale audio in its factories. Flushing them both [Aug 18 10:34:07] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP ooh, format changed from none to ulaw [Aug 18 10:34:07] DEBUG[13978] stasis/app.c: Channel '213081' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005a - start 1629282840.742909 answer 0.000000 end 1629282847.011957 dur 6.269 bill 1629282847.011 dispo NO ANSWER [Aug 18 10:34:07] DEBUG[14112] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[13978] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13978] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:07] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:07] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 944, ms is 79 [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13666] audiohook.c: Audiohook 0x7f0c80063290 has stale audio in its factories. Flushing them both [Aug 18 10:34:07] DEBUG[13683] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14112] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14112] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as2aed188c Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" Content-Length: 0 <-------------> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:07] DEBUG[14112] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1cecb88b;received=159.65.48.104 [Aug 18 10:34:07] DEBUG[14112] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 is already using the new technology. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:07] DEBUG[14112] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c2c006860(SIP/zvonobot-00000033) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2aed188c [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14116] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pushing 0x7f0c20083a40(SIP/zvonobot-00000004) [Aug 18 10:34:07] DEBUG[13872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[14115] channel.c: Channel Announcer/ARI-00000027;1 setting write format path: gsm -> slin [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTP changing ssrc from 231714101 to 1621973554 due to a source change [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd640e9" [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as2aed188c [Aug 18 10:34:07] DEBUG[14093] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[13967] channel.c: Channel 0x7f0c9c08d3f0 'SIP/zvonobot-00000075' allocated [Aug 18 10:34:07] DEBUG[14093] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:07] DEBUG[13967] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Outgoing Call for 79821116959 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Stopping retransmission on '596ef69a675656833620d8345b47f33c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] DEBUG[13263] stasis/app.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' is 2 interested in calls_0 [Aug 18 10:34:07] DEBUG[14118] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14118] http.c: HTTP Request URI is /ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/record?name=213015_dBotEnLtTVjDPxFVWfqmvQfKraUbAFtr&format=wav [Aug 18 10:34:07] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116992@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf Max-Forwards: 70 From: ;tag=as57de5f2f To: Contact: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:07] DEBUG[13967] res_stasis.c: calls_0: Subscribing to 213077 [Aug 18 10:34:07] VERBOSE[14116] bridge_channel.c: Channel SIP/zvonobot-00000004 joined 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[13967] stasis/app.c: Channel '213077' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Outgoing Call for 79821116963 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] DEBUG[13967] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13967] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:07] DEBUG[14118] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/record] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[14115] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] VERBOSE[14115] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14118] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/record] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:07] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:07] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:07] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14116] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[14116] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[14116] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14116] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14116] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[14116] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:07] DEBUG[14116] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c20083a40(SIP/zvonobot-00000004) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 736, ms is 66 [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:07] VERBOSE[14119] chan_sip.c: Audio is at 14548 [Aug 18 10:34:07] VERBOSE[14119] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:07] VERBOSE[14117] chan_sip.c: Audio is at 11206 [Aug 18 10:34:07] VERBOSE[14117] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:07] VERBOSE[14117] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:07] VERBOSE[14117] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:07] DEBUG[14116] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTP changing ssrc from 1364366715 to 1096189093 due to a source change [Aug 18 10:34:07] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[12888] stasis/app.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' is 2 interested in calls_0 [Aug 18 10:34:07] DEBUG[14107] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[14107] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14120] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 640, ms is 60 [Aug 18 10:34:07] DEBUG[14120] http.c: HTTP Request URI is /ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/record?name=212967_lhjsPNnYXjMyUoluhutJnnhSQwIrUluO&format=wav [Aug 18 10:34:07] DEBUG[14118] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/record] with handler [ari] len 3 [Aug 18 10:34:07] VERBOSE[14119] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14120] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/record] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Initializing initreq for method INVITE - callid 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116959@178.62.121.41 SIP/2.0 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 3 [ 52]: From: ;tag=as3ecc0b7c [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14118] http.c: Match made with [ari] [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 From: ;tag=as52d5bd88 To: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 6 [ 60]: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (5) INVITE - 5 [Aug 18 10:34:07] VERBOSE[14119] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[14120] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/record] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116972@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a Max-Forwards: 70 From: ;tag=as1ed67fff To: Contact: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 476472901 476472901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:07 GMT [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:07] VERBOSE[14117] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Finding handler for bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/record [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #107 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116991@178.62.121.41", nonce="416821cf", response="3c75583eb1ebe6596f83638c5014d5c4" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 698668828 698668829 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116994@178.62.121.41", nonce="72646695", response="b512504387f54cbeaa8c8fb845d86e20" Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 875682952 875682953 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14120] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/record] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 From: ;tag=as009e460a To: ;tag=as64de9d5c Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 568221000 568221000 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13198 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as009e460a [Aug 18 10:34:07] DEBUG[14120] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[13982] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13982] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13982] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13982] channel.c: Channel Announcer/ARI-00000026;1 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64de9d5c [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:07] DEBUG[13982] channel.c: Channel 0x7f0c280a4fb0 'Announcer/ARI-00000026;1' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:07] DEBUG[14117] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Initializing initreq for method INVITE - callid 564726e17074235c1af6801638e43e42@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116963@178.62.121.41 SIP/2.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Finding handler for bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Finding handler for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14118] res_ari.c: No explicit handler found for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Finding handler for record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:07] DEBUG[14118] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:07] DEBUG[14118] stasis.c: Creating topic. name: channel:1629282847.266, detail: [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Finding handler for 9cefb3ad-33ea-4a52-96a1-42b677d6802c [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14120] res_ari.c: No explicit handler found for 9cefb3ad-33ea-4a52-96a1-42b677d6802c. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Finding handler for record [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:07] DEBUG[14118] stasis.c: Topic 'channel:1629282847.266': 0x7f0c7804a270 created [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:07] VERBOSE[14117] dial.c: Called zvonobot/79821116959 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:07] DEBUG[13979] channel.c: Channel 0x7f0cb4080db0 'SIP/zvonobot-00000076' allocated [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:07] DEBUG[13979] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 3 [ 52]: From: ;tag=as27ec27d0 [Aug 18 10:34:07] DEBUG[13979] res_stasis.c: calls_0: Subscribing to 213083 [Aug 18 10:34:07] DEBUG[13979] stasis/app.c: Channel '213083' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 568221000 568221000 IN IP4 178.62.121.41 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13198 RTP/AVP 0 8 101 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 (Checking To) --From tag as009e460a --To-tag as64de9d5c [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Stopping retransmission on '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Got SDP version 568221000 and unique parts [root 568221000 IN IP4 178.62.121.41] [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 568221000 568221000 IN IP4 178.62.121.41... OK. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:07] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:07] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:07] DEBUG[13979] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13979] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38043ba0) ICE set role failed; no ice instance [Aug 18 10:34:07] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38043ba0) RTCP setting address on RTP instance [Aug 18 10:34:07] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c380456f0 -- Strict RTP learning after remote address set to: 178.62.121.41:13198 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:13198 [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00250e8) from 0x7f0c147e2330 to 0x7f0c38043d78 [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb008d398) from 0x7f0c147e2330 to 0x7f0c38043d78 [Aug 18 10:34:07] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb008d3e8) from 0x7f0c147e2330 to 0x7f0c38043d78 [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38043ba0) RTCP ignoring duplicate property [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:07] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000038 setting read format path: alaw -> alaw [Aug 18 10:34:07] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000038 setting write format path: alaw -> alaw [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38043ba0) DTLS - ast_rtp_activate rtp=0x7f0c380456f0 - setup and perform DTLS' [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c380456f0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c380456f0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:07] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:07] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Outgoing Call for 79821116957 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:07] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP got report of 100 bytes from 178.62.121.41:11671 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:07] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:07] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117018@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4dcfccf7 Max-Forwards: 70 From: ;tag=as009e460a To: ;tag=as64de9d5c Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[14118] stasis.c: Creating topic. name: cache:311/channel:1629282847.266, detail: [Aug 18 10:34:07] DEBUG[14118] stasis.c: Topic 'cache:311/channel:1629282847.266': 0x7f0c780344c0 created [Aug 18 10:34:07] VERBOSE[13286] dial.c: SIP/zvonobot-00000038 answered [Aug 18 10:34:07] VERBOSE[13286] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000038 [Aug 18 10:34:07] DEBUG[14120] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 6 [ 60]: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #92 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #92)) [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:07 GMT [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:07] VERBOSE[14119] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #100 [Aug 18 10:34:07] DEBUG[14119] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[13286] stasis/app.c: Channel '213022' is 2 interested in calls_0 [Aug 18 10:34:07] DEBUG[14120] stasis.c: Creating topic. name: channel:1629282847.267, detail: [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116973@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594 Max-Forwards: 70 From: ;tag=as6d27c109 To: Contact: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 837232621 837232621 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11962 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116969@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8 Max-Forwards: 70 From: ;tag=as2f5156ef To: Contact: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1585483767 1585483767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6236ms with no response [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Hanging up call 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[13772] channel.c: Channel 0x7f0c280d3fe0 'SIP/zvonobot-0000005b' hanging up. Refs: 2 [Aug 18 10:34:07] VERBOSE[14119] dial.c: Called zvonobot/79821116963 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Session timer started: 51 - 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 1768000ms [Aug 18 10:34:07] VERBOSE[14122] chan_sip.c: Audio is at 18420 [Aug 18 10:34:07] VERBOSE[14122] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:07] VERBOSE[14122] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:07] VERBOSE[14122] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660 Max-Forwards: 70 From: ;tag=as6657c8e8 To: ;tag=as510b84fe Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="589c1c16", response="87884a26a92648e14764df9a3df9354f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005b - start 1629282840.858672 answer 0.000000 end 1629282847.323970 dur 6.465 bill 1629282847.323 dispo NO ANSWER [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Initializing initreq for method INVITE - callid 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116957@178.62.121.41 SIP/2.0 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:07] DEBUG[14123] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14123] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:07] DEBUG[14123] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[14123] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[14120] stasis.c: Topic 'channel:1629282847.267': 0x7f0c8002b2b0 created [Aug 18 10:34:07] DEBUG[14123] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[14123] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 3 [ 52]: From: ;tag=as4bf88154 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6657c8e8 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as510b84fe [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="589c1c16", response="87884a26a92648e14764df9a3df9354f" [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 (Checking From) --From tag as6657c8e8 --To-tag as510b84fe [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 6 [ 60]: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:07] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:07] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:07 GMT [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[14120] stasis.c: Creating topic. name: cache:312/channel:1629282847.267, detail: [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:07] VERBOSE[14122] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #104 [Aug 18 10:34:07] DEBUG[14122] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14123] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14120] stasis.c: Topic 'cache:312/channel:1629282847.267': 0x7f0c80030150 created [Aug 18 10:34:07] VERBOSE[14122] dial.c: Called zvonobot/79821116957 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14123] stasis.c: Creating topic. name: bridge:fbfe71c6-df7c-4b8c-8b66-37207aaf9846, detail: [Aug 18 10:34:07] DEBUG[14123] stasis.c: Topic 'bridge:fbfe71c6-df7c-4b8c-8b66-37207aaf9846': 0x7f0c88038740 created [Aug 18 10:34:07] DEBUG[14123] stasis.c: Creating topic. name: cache:313/bridge:fbfe71c6-df7c-4b8c-8b66-37207aaf9846, detail: [Aug 18 10:34:07] DEBUG[14123] stasis.c: Topic 'cache:313/bridge:fbfe71c6-df7c-4b8c-8b66-37207aaf9846': 0x7f0c88078da0 created [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14123] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[14123] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[14123] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14123] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:07] DEBUG[14123] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[14123] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: calling simple_bridge technology constructor [Aug 18 10:34:07] DEBUG[14123] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: calling simple_bridge technology start [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP audio difference is 696, ms is 107 [Aug 18 10:34:07] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 768, ms is 116 [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13465] channel.c: Channel 0x7f0c7c00f100 'Recorder/ARI-00000013;1' destroying [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:07] DEBUG[13761] channel.c: Channel 0x7f0c0806c2f0 'Announcer/ARI-00000022;2' destroying [Aug 18 10:34:07] DEBUG[13760] channel.c: Channel 0x7f0c90044400 'Announcer/ARI-00000021;2' destroying [Aug 18 10:34:07] DEBUG[13474] channel.c: Channel 0x7f0c98012a90 'SIP/zvonobot-00000023' destroying [Aug 18 10:34:07] DEBUG[13867] channel.c: Channel 0x7f0c88099000 'Snoop/212982-00000010' allocated [Aug 18 10:34:07] DEBUG[13900] channel.c: Channel 0x7f0c1c13de00 'Announcer/ARI-00000028;2' allocated [Aug 18 10:34:07] DEBUG[13760] stasis.c: Destroying topic. name: cache:228/channel:1629282840.191, detail: [Aug 18 10:34:07] DEBUG[13704] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677'. Checking compatability for channels 'SIP/zvonobot-00000013' and 'Recorder/ARI-00000024;2' [Aug 18 10:34:07] DEBUG[13704] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as channel 'SIP/zvonobot-00000013' has features which prevent it [Aug 18 10:34:07] DEBUG[13704] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[13704] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13704] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13704] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[13704] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677 is already using the new technology. [Aug 18 10:34:07] DEBUG[14123] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13900] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:07] DEBUG[13900] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000028;1' [Aug 18 10:34:07] DEBUG[13760] stasis.c: Topic 'cache:228/channel:1629282840.191': 0x7f0c9005c640 destroyed [Aug 18 10:34:07] DEBUG[13760] stasis.c: Destroying topic. name: channel:1629282840.191, detail: [Aug 18 10:34:07] DEBUG[13760] stasis.c: Topic 'channel:1629282840.191': 0x7f0c90062c40 destroyed [Aug 18 10:34:07] DEBUG[14123] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14131] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14136] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13867] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:07] DEBUG[14131] http.c: HTTP Request URI is /ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/addChannel?channel=213022 [Aug 18 10:34:07] DEBUG[14136] http.c: HTTP Request URI is /ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play?media=sound%3Asilence%2F2 [Aug 18 10:34:07] DEBUG[14139] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14131] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[14131] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[14136] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[14131] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/addChannel] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[13867] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14136] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[14139] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212982&app=calls_0&format=slin16&external_host=127.0.0.1%3A50497 [Aug 18 10:34:07] DEBUG[14131] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[14136] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[14136] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[14139] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282847.268, detail: [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'channel:1629282847.268': 0x7f0c30023840 created [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Finding handler for bridges/45640e14-e267-477d-81ea-fbac374f9677/play [Aug 18 10:34:07] DEBUG[14139] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: channel '212999': is 0 interested in calls_0 [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: channel '212999' unsubscribed from calls_0 [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Finding handler for bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/addChannel [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[20545] stasis.c: Creating topic. name: cache:314/channel:1629282847.268, detail: [Aug 18 10:34:07] DEBUG[14133] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c1c136810(Announcer/ARI-00000028;2) is joining [Aug 18 10:34:07] DEBUG[13761] stasis.c: Destroying topic. name: cache:229/channel:1629282840.192, detail: [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14139] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[13761] stasis.c: Topic 'cache:229/channel:1629282840.192': 0x7f0c0804fc30 destroyed [Aug 18 10:34:07] DEBUG[13761] stasis.c: Destroying topic. name: channel:1629282840.192, detail: [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[14139] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'cache:314/channel:1629282847.268': 0x7f0c300ba000 created [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Finding handler for fbfe71c6-df7c-4b8c-8b66-37207aaf9846 [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Finding handler for 45640e14-e267-477d-81ea-fbac374f9677 [Aug 18 10:34:07] DEBUG[20545] stasis.c: Destroying topic. name: cache:314/channel:1629282847.268, detail: [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'cache:314/channel:1629282847.268': 0x7f0c300ba000 destroyed [Aug 18 10:34:07] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282847.268, detail: [Aug 18 10:34:07] DEBUG[13761] stasis.c: Topic 'channel:1629282840.192': 0x7f0c08086490 destroyed [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14136] res_ari.c: No explicit handler found for 45640e14-e267-477d-81ea-fbac374f9677. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14131] res_ari.c: No explicit handler found for fbfe71c6-df7c-4b8c-8b66-37207aaf9846. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Finding handler for play [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Finding handler for addChannel [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'channel:1629282847.268': 0x7f0c30023840 destroyed [Aug 18 10:34:07] DEBUG[14131] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:07] DEBUG[13976] channel.c: Channel 0x7f0cb0160ed0 'SIP/zvonobot-00000077' allocated [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:07] DEBUG[13976] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Finding handler for channels [Aug 18 10:34:07] DEBUG[13465] stasis.c: Destroying topic. name: cache:153/channel:1629282835.128, detail: [Aug 18 10:34:07] DEBUG[13465] stasis.c: Topic 'cache:153/channel:1629282835.128': 0x7f0c7c010d50 destroyed [Aug 18 10:34:07] DEBUG[13465] stasis.c: Destroying topic. name: channel:1629282835.128, detail: [Aug 18 10:34:07] DEBUG[13465] stasis.c: Topic 'channel:1629282835.128': 0x7f0c7c01ae00 destroyed [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:07] DEBUG[14131] stasis/control.c: 213022: Sending channel add_to_bridge command [Aug 18 10:34:07] DEBUG[14136] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[13474] stasis.c: Destroying topic. name: cache:51/channel:212999, detail: [Aug 18 10:34:07] DEBUG[14136] stasis.c: Creating topic. name: channel:1629282847.269, detail: [Aug 18 10:34:07] DEBUG[14136] stasis.c: Topic 'channel:1629282847.269': 0x7f0c9c03a280 created [Aug 18 10:34:07] DEBUG[14136] stasis.c: Creating topic. name: cache:315/channel:1629282847.269, detail: [Aug 18 10:34:07] DEBUG[14136] stasis.c: Topic 'cache:315/channel:1629282847.269': 0x7f0c9c06dcb0 created [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:07] DEBUG[13474] stasis.c: Topic 'cache:51/channel:212999': 0x7f0c98078800 destroyed [Aug 18 10:34:07] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:48', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000023', '', 'Stasis', 'calls_0', 15, 8, 'ANSWERED', 3, '', '212999', '')] [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:07] DEBUG[13459] res_rtp_asterisk.c: (0x7f0c94022610) RTP no remote address on instance, so dropping frame [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:07] DEBUG[13474] stasis.c: Destroying topic. name: channel:212999, detail: [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:07] DEBUG[13474] stasis.c: Topic 'channel:212999': 0x7f0c980793e0 destroyed [Aug 18 10:34:07] DEBUG[14139] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:07] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 840, ms is 125 [Aug 18 10:34:07] DEBUG[14139] netsock2.c: Splitting '127.0.0.1:50497' into... [Aug 18 10:34:07] DEBUG[13474] channel.c: Channel 0x7f0ca0056680 'Snoop/212999-00000008' destroying [Aug 18 10:34:07] DEBUG[14139] netsock2.c: ...host '127.0.0.1' and port '50497'. [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[13976] res_stasis.c: calls_0: Subscribing to 213080 [Aug 18 10:34:07] DEBUG[13976] stasis/app.c: Channel '213080' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[13459] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-0000002e' [Aug 18 10:34:07] DEBUG[14139] netsock2.c: Splitting '127.0.0.1:50497' into... [Aug 18 10:34:07] DEBUG[14139] netsock2.c: ...host '127.0.0.1' and port '50497'. [Aug 18 10:34:07] DEBUG[14139] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:07] DEBUG[14139] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c98083570' [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) RTP allocated port 12248 [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) ICE creating session 127.0.0.1:12248 (12248) [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) ICE create [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) ICE add system candidates [Aug 18 10:34:07] DEBUG[14139] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:07] DEBUG[14139] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) ICE add candidate: 159.65.48.104:12248, 2130706431 [Aug 18 10:34:07] DEBUG[14139] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:07] DEBUG[14139] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Outgoing Call for 79821116960 [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:07] DEBUG[13888] channel.c: Channel 0x7f0c8408c060 'Announcer/ARI-00000025;1' destroying [Aug 18 10:34:07] DEBUG[14139] res_rtp_asterisk.c: (0x7f0c98083570) ICE add candidate: 10.131.0.10:12248, 2130706431 [Aug 18 10:34:07] DEBUG[13942] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:07] DEBUG[13945] channel.c: Channel 0x7f0c78095bd0 'Recorder/ARI-00000029;2' allocated [Aug 18 10:34:07] DEBUG[13945] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:07] DEBUG[14140] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14139] rtp_engine.c: RTP instance '0x7f0c98083570' is setup and ready to go [Aug 18 10:34:07] DEBUG[14139] stasis.c: Creating topic. name: channel:robot_212982, detail: [Aug 18 10:34:07] DEBUG[14139] stasis.c: Topic 'channel:robot_212982': 0x7f0c9803bc20 created [Aug 18 10:34:07] DEBUG[14139] stasis.c: Creating topic. name: cache:316/channel:robot_212982, detail: [Aug 18 10:34:07] DEBUG[14139] stasis.c: Topic 'cache:316/channel:robot_212982': 0x7f0c980786f0 created [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:07] VERBOSE[14141] chan_sip.c: Audio is at 15846 [Aug 18 10:34:07] VERBOSE[14141] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:07] VERBOSE[14141] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:07] VERBOSE[14141] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:07] DEBUG[13976] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13976] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14133] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pushing 0x7f0c1c136810(Announcer/ARI-00000028;2) [Aug 18 10:34:07] DEBUG[14140] http.c: HTTP Request URI is /ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:34:07] DEBUG[13942] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14140] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[13888] stasis.c: Destroying topic. name: cache:249/channel:1629282841.210, detail: [Aug 18 10:34:07] DEBUG[13888] stasis.c: Topic 'cache:249/channel:1629282841.210': 0x7f0c84101300 destroyed [Aug 18 10:34:07] DEBUG[13888] stasis.c: Destroying topic. name: channel:1629282841.210, detail: [Aug 18 10:34:07] DEBUG[13888] stasis.c: Topic 'channel:1629282841.210': 0x7f0c8408a4f0 destroyed [Aug 18 10:34:07] DEBUG[13704] audiohook.c: Audiohook 0x7f0c8805b620 has stale audio in its factories. Flushing them both [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Initializing initreq for method INVITE - callid 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14143] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c7804b2f0(Recorder/ARI-00000029;2) is joining [Aug 18 10:34:07] DEBUG[13866] bridge_channel.c: Setting 0x7f0c8408a2a0(Announcer/ARI-00000025;2) state from:0 to:1 [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14140] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[14140] http.c: match request [ari/bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[13866] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pulling 0x7f0c8408a2a0(Announcer/ARI-00000025;2) [Aug 18 10:34:07] DEBUG[14140] http.c: Match made with [ari] [Aug 18 10:34:07] VERBOSE[13866] bridge_channel.c: Channel Announcer/ARI-00000025;2 left 'softmix' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116960@178.62.121.41 SIP/2.0 [Aug 18 10:34:07] DEBUG[13866] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c8408a2a0(Announcer/ARI-00000025;2) is leaving softmix technology [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660;received=178.62.121.41 From: ;tag=as6657c8e8 To: ;tag=as510b84fe Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Finding handler for bridges/85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 [Aug 18 10:34:07] DEBUG[13866] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee'. Checking compatability for channels 'SIP/zvonobot-0000003b' and 'Recorder/ARI-0000001e;2' [Aug 18 10:34:07] DEBUG[13866] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as channel 'SIP/zvonobot-0000003b' has features which prevent it [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13866] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[14132] stasis/app.c: Channel '1629282845.251' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 3 [ 52]: From: ;tag=as1180a433 [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Finding handler for 85c47f7b-0e24-4408-b7bf-5a532802bd8e [Aug 18 10:34:07] VERBOSE[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: switching from softmix technology to simple_bridge [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology constructor [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0ca803dbf0(SIP/zvonobot-0000003b) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0c94055bb0(Recorder/ARI-0000001e;2) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is leaving softmix technology (dummy) [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is leaving softmix technology (dummy) [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology stop [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[13866] channel.c: Channel Recorder/ARI-0000001e;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13866] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[13866] channel.c: Channel Recorder/ARI-0000001e;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13866] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology start [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: deferring softmix technology destructor [Aug 18 10:34:07] DEBUG[13866] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: queueing action type:13 sub:1000 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 6 [ 60]: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[14140] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14143] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pushing 0x7f0c7804b2f0(Recorder/ARI-00000029;2) [Aug 18 10:34:07] DEBUG[14140] res_ari.c: No explicit handler found for 85c47f7b-0e24-4408-b7bf-5a532802bd8e. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14132] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 100 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[13474] stasis.c: Destroying topic. name: cache:159/channel:1629282835.132, detail: [Aug 18 10:34:07] DEBUG[14140] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: telling all channels to leave the party [Aug 18 10:34:07] DEBUG[14140] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:07] DEBUG[13474] stasis.c: Topic 'cache:159/channel:1629282835.132': 0x7f0ca0058660 destroyed [Aug 18 10:34:07] DEBUG[14140] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: queueing action type:13 sub:1001 [Aug 18 10:34:07] DEBUG[13474] stasis.c: Destroying topic. name: channel:1629282835.132, detail: [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:07 GMT [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:07] VERBOSE[14141] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #109 [Aug 18 10:34:07] DEBUG[14141] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14140] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[14140] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282847.271, detail: [Aug 18 10:34:07] DEBUG[13474] stasis.c: Topic 'channel:1629282835.132': 0x7f0ca0058450 destroyed [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'channel:1629282847.271': 0x7f0c300ba000 created [Aug 18 10:34:07] DEBUG[20545] stasis.c: Creating topic. name: cache:317/channel:1629282847.271, detail: [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'cache:317/channel:1629282847.271': 0x7f0c30023840 created [Aug 18 10:34:07] DEBUG[20545] stasis.c: Destroying topic. name: cache:317/channel:1629282847.271, detail: [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'cache:317/channel:1629282847.271': 0x7f0c30023840 destroyed [Aug 18 10:34:07] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282847.271, detail: [Aug 18 10:34:07] DEBUG[20545] stasis.c: Topic 'channel:1629282847.271': 0x7f0c300ba000 destroyed [Aug 18 10:34:07] VERBOSE[14141] dial.c: Called zvonobot/79821116960 [Aug 18 10:34:07] DEBUG[14144] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (1) INVITE - 5 [Aug 18 10:34:07] DEBUG[14144] http.c: HTTP Request URI is /ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #111 (3) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #111)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14144] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Setting 0x7f0c9006b170(SIP/zvonobot-0000002e) state from:0 to:1 [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pulling 0x7f0c9006b170(SIP/zvonobot-0000002e) [Aug 18 10:34:07] VERBOSE[13459] bridge_channel.c: Channel SIP/zvonobot-0000002e left 'softmix' stasis-bridge [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9006b170(SIP/zvonobot-0000002e) is leaving softmix technology [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Setting 0x7f0c9c048790(Announcer/ARI-0000001d;2) state from:0 to:2 [Aug 18 10:34:07] DEBUG[13459] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267'. Checking compatability for channels 'Announcer/ARI-0000001d;2' and 'Recorder/ARI-00000014;2' [Aug 18 10:34:07] DEBUG[13459] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as could not get details [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13459] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] VERBOSE[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: switching from softmix technology to simple_bridge [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology constructor [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c9c048790(Announcer/ARI-0000001d;2) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: moving 0x7f0c8808e340(Recorder/ARI-00000014;2) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9c048790(Announcer/ARI-0000001d;2) is leaving softmix technology (dummy) [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is leaving softmix technology (dummy) [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology stop [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9c048790(Announcer/ARI-0000001d;2) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Recorder/ARI-00000014;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Announcer/ARI-0000001d;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Announcer/ARI-0000001d;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Recorder/ARI-00000014;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Announcer/ARI-0000001d;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Announcer/ARI-0000001d;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel Recorder/ARI-00000014;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling simple_bridge technology start [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: deferring softmix technology destructor [Aug 18 10:34:07] DEBUG[13459] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: queueing action type:13 sub:1000 [Aug 18 10:34:07] DEBUG[14143] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:07] VERBOSE[14143] bridge_channel.c: Channel Recorder/ARI-00000029;2 joined 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (1) INVITE - 5 [Aug 18 10:34:07] DEBUG[14144] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:55', '"" <>', '', 's', 'default', 'Snoop/212999-00000008', 'UnicastRTP/127.0.0.1:50116-0x7f0c24077280', 'Stasis', 'calls_0', 6, 6, 'ANSWERED', 3, '', '1629282835.132', '')] [Aug 18 10:34:07] DEBUG[14144] http.c: match request [ari/bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[14144] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Finding handler for bridges/6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Finding handler for bridges [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:07] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP audio difference is 736, ms is 112 [Aug 18 10:34:07] DEBUG[13872] bridge_softmix.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: stopping mixing thread [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:07] DEBUG[20534] bridge_softmix.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: Waiting for mixing thread to die. [Aug 18 10:34:07] DEBUG[13627] channel.c: SIP/zvonobot-0000003b: Dropping redundant connected line update "" <>. [Aug 18 10:34:07] DEBUG[13671] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pulling 0x7f0c9c048790(Announcer/ARI-0000001d;2) [Aug 18 10:34:07] DEBUG[13679] channel.c: Recorder/ARI-0000001e;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:07] DEBUG[13286] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000038 [Aug 18 10:34:07] DEBUG[13286] stasis/control.c: 213022: Adding to bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 [Aug 18 10:34:07] DEBUG[13286] stasis/app.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:07] DEBUG[14146] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c88072950(SIP/zvonobot-00000038) is joining [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling stasis bridge destructor [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology stop [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 85c47f7b-0e24-4408-b7bf-5a532802bd8e: calling simple_bridge technology destructor [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14054] channel.c: Channel 0x7f0c940b1960 'SIP/zvonobot-00000078' allocated [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:07] DEBUG[14054] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (1) INVITE - 5 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:07] DEBUG[14132] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14132] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e': is 0 interested in calls_0 [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: bridge '85c47f7b-0e24-4408-b7bf-5a532802bd8e' unsubscribed from calls_0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] VERBOSE[13671] bridge_channel.c: Channel Announcer/ARI-0000001d;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:07] DEBUG[13671] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c9c048790(Announcer/ARI-0000001d;2) is leaving simple_bridge technology [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:07] DEBUG[13671] bridge_channel.c: Setting 0x7f0c8808e340(Recorder/ARI-00000014;2) state from:0 to:2 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:07] DEBUG[20534] stasis.c: Destroying topic. name: cache:108/bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e, detail: [Aug 18 10:34:07] DEBUG[14143] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b'. Checking compatability for channels 'SIP/zvonobot-0000001c' and 'Recorder/ARI-00000029;2' [Aug 18 10:34:07] DEBUG[20534] stasis.c: Topic 'cache:108/bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e': 0x7f0c7801a060 destroyed [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000002e - start 1629282830.167186 answer 1629282835.189588 end 1629282847.691803 dur 17.524 bill 12.502 dispo ANSWERED [Aug 18 10:34:07] DEBUG[14133] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:07] DEBUG[14146] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: pushing 0x7f0c88072950(SIP/zvonobot-00000038) [Aug 18 10:34:07] DEBUG[13866] channel.c: Channel 0x7f0c8410b600 'Announcer/ARI-00000025;2' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[20534] stasis.c: Destroying topic. name: bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e, detail: [Aug 18 10:34:07] DEBUG[20534] stasis.c: Topic 'bridge:85c47f7b-0e24-4408-b7bf-5a532802bd8e': 0x7f0c7802d570 destroyed [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Finding handler for 6f6fd705-00c9-4b9f-a75f-d7c933b85e66 [Aug 18 10:34:07] DEBUG[14143] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as could not get details [Aug 18 10:34:07] VERBOSE[14133] bridge_channel.c: Channel Announcer/ARI-00000028;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:07] DEBUG[14144] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Session timer stopped: 22 - 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14144] res_ari.c: No explicit handler found for 6f6fd705-00c9-4b9f-a75f-d7c933b85e66. Using wildcard bridgeId. [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:07] DEBUG[14143] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] VERBOSE[14146] bridge_channel.c: Channel SIP/zvonobot-00000038 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:07] DEBUG[14143] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Bridge is returning 0x7f0c9006b170(SIP/zvonobot-0000002e) to read format alaw [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel SIP/zvonobot-0000002e setting read format path: ulaw -> alaw [Aug 18 10:34:07] DEBUG[13459] bridge_channel.c: Bridge is returning 0x7f0c9006b170(SIP/zvonobot-0000002e) to write format alaw [Aug 18 10:34:07] DEBUG[13459] channel.c: Channel SIP/zvonobot-0000002e setting write format path: alaw -> ulaw [Aug 18 10:34:07] DEBUG[13459] stasis/control.c: 213012, e2e70698-2279-429d-a48c-2fe9dd817267: Channel was departed from bridge [Aug 18 10:34:07] DEBUG[13459] stasis/app.c: bridge 'e2e70698-2279-429d-a48c-2fe9dd817267': is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[13195] stasis/control.c: 213012: Channel departing bridge [Aug 18 10:34:07] DEBUG[13195] bridge.c: Waiting for 0x7f0c9006b170(SIP/zvonobot-0000002e) bridge thread to die. [Aug 18 10:34:07] DEBUG[13459] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:07] DEBUG[13195] stasis/app.c: channel '213012': is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[14144] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: telling all channels to leave the party [Aug 18 10:34:07] DEBUG[13195] channel.c: Channel 0x7f0c94027db0 'SIP/zvonobot-0000002e' hanging up. Refs: 3 [Aug 18 10:34:07] DEBUG[14143] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14143] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[14143] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:07] DEBUG[14144] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[14143] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c7804b2f0(Recorder/ARI-00000029;2) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:07] DEBUG[13671] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[13671] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[13671] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13671] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[13671] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[13671] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267 is already using the new technology. [Aug 18 10:34:07] DEBUG[14144] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: queueing action type:13 sub:1001 [Aug 18 10:34:07] DEBUG[14144] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[13680] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: stopping mixing thread [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:07] DEBUG[13468] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: pulling 0x7f0c8808e340(Recorder/ARI-00000014;2) [Aug 18 10:34:07] VERBOSE[13468] bridge_channel.c: Channel Recorder/ARI-00000014;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:07] DEBUG[13468] bridge_channel.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: 0x7f0c8808e340(Recorder/ARI-00000014;2) is leaving simple_bridge technology [Aug 18 10:34:07] DEBUG[14144] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14143] channel.c: Channel Recorder/ARI-00000029;2 setting read format path: slin -> slin [Aug 18 10:34:07] DEBUG[20534] bridge_softmix.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267: Waiting for mixing thread to die. [Aug 18 10:34:07] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14148] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14148] http.c: HTTP Request URI is /ari/playbacks/354c955c-8a71-439e-92e6-35f1bf612218 [Aug 18 10:34:07] DEBUG[13468] bridge_native_rtp.c: Bridge 'e2e70698-2279-429d-a48c-2fe9dd817267' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[14143] channel.c: Channel SIP/zvonobot-0000001c setting write format path: slin -> ulaw [Aug 18 10:34:07] DEBUG[13468] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[13468] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14148] http.c: match request [ari/playbacks/354c955c-8a71-439e-92e6-35f1bf612218] with handler [httpstatus] len 10 [Aug 18 10:34:07] DEBUG[13468] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14148] http.c: match request [ari/playbacks/354c955c-8a71-439e-92e6-35f1bf612218] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[13468] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[13468] bridge.c: Bridge e2e70698-2279-429d-a48c-2fe9dd817267 is already using the new technology. [Aug 18 10:34:07] DEBUG[14143] channel.c: Channel SIP/zvonobot-0000001c setting read format path: ulaw -> slin [Aug 18 10:34:07] DEBUG[14063] channel.c: Channel 0x7f0c9c09ffc0 'SIP/zvonobot-00000079' allocated [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:07] DEBUG[14063] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:07] DEBUG[13884] channel.c: Channel 0x7f0ca4085510 'Announcer/ARI-0000002a;1' allocated [Aug 18 10:34:07] DEBUG[13884] stasis.c: Creating topic. name: channel:1629282847.272, detail: [Aug 18 10:34:07] DEBUG[13884] stasis.c: Topic 'channel:1629282847.272': 0x7f0ca4059600 created [Aug 18 10:34:07] DEBUG[13884] stasis.c: Creating topic. name: cache:318/channel:1629282847.272, detail: [Aug 18 10:34:07] DEBUG[13884] stasis.c: Topic 'cache:318/channel:1629282847.272': 0x7f0ca40597d0 created [Aug 18 10:34:07] DEBUG[13884] channel.c: Channel 0x7f0ca405f210 'Announcer/ARI-0000002a;2' allocated [Aug 18 10:34:07] DEBUG[13884] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:07] DEBUG[13884] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000002a;1' [Aug 18 10:34:07] DEBUG[14148] http.c: match request [ari/playbacks/354c955c-8a71-439e-92e6-35f1bf612218] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[14146] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as two channels are required [Aug 18 10:34:07] DEBUG[14146] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:07] DEBUG[14143] channel.c: Channel Recorder/ARI-00000029;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[14150] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0ca403e270(Announcer/ARI-0000002a;2) is joining [Aug 18 10:34:07] DEBUG[14148] http.c: Match made with [ari] [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Finding handler for playbacks/354c955c-8a71-439e-92e6-35f1bf612218 [Aug 18 10:34:07] DEBUG[14146] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[14146] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Finding handler for playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Finding handler for 354c955c-8a71-439e-92e6-35f1bf612218 [Aug 18 10:34:07] DEBUG[14148] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:07] DEBUG[14148] res_ari.c: No explicit handler found for 354c955c-8a71-439e-92e6-35f1bf612218. Using wildcard playbackId. [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14133] bridge.c: Chose bridge technology softmix [Aug 18 10:34:07] VERBOSE[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: switching from simple_bridge technology to softmix [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology constructor [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: calling stasis bridge destructor [Aug 18 10:34:07] DEBUG[14148] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: calling simple_bridge technology stop [Aug 18 10:34:07] DEBUG[20534] bridge.c: Bridge 6f6fd705-00c9-4b9f-a75f-d7c933b85e66: calling simple_bridge technology destructor [Aug 18 10:34:07] DEBUG[14148] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14146] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:07] DEBUG[13683] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13683] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:07] DEBUG[13683] channel.c: Channel Announcer/ARI-0000001d;1 setting write format path: slin -> slin [Aug 18 10:34:07] NOTICE[13683] res_stasis_playback.c: 1629282838.171: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:07] DEBUG[13683] channel.c: Channel 0x7f0c9c024160 'Announcer/ARI-0000001d;1' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[14146] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:07] DEBUG[13671] channel.c: Channel 0x7f0c9c05ac20 'Announcer/ARI-0000001d;2' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66': is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[20620] stasis/app.c: bridge '6f6fd705-00c9-4b9f-a75f-d7c933b85e66' unsubscribed from calls_0 [Aug 18 10:34:07] DEBUG[13468] channel.c: Channel 0x7f0c88086180 'Recorder/ARI-00000014;2' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[14150] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pushing 0x7f0ca403e270(Announcer/ARI-0000002a;2) [Aug 18 10:34:07] DEBUG[14146] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c88072950(SIP/zvonobot-00000038) is joining simple_bridge technology [Aug 18 10:34:07] DEBUG[20620] stasis.c: Destroying topic. name: cache:120/bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66, detail: [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0ca0073e00(SIP/zvonobot-00000030) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[20620] stasis.c: Topic 'cache:120/bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66': 0x7f0c3803d560 destroyed [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0c78074930(Recorder/ARI-0000001c;2) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is leaving simple_bridge technology (dummy) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology stop [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c1c136810(Announcer/ARI-00000028;2) is joining softmix technology [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Announcer/ARI-00000028;2: [Aug 18 10:34:07] DEBUG[14133] channel.c: Channel Announcer/ARI-00000028;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Announcer/ARI-00000028;2: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Announcer/ARI-00000028;2: Not in SFU mode [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is joining softmix technology [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: SIP/zvonobot-00000030: [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: SIP/zvonobot-00000030: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: SIP/zvonobot-00000030: Not in SFU mode [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is joining softmix technology [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Recorder/ARI-0000001c;2: [Aug 18 10:34:07] DEBUG[14133] channel.c: Channel Recorder/ARI-0000001c;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14133] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Recorder/ARI-0000001c;2: [Aug 18 10:34:07] DEBUG[14133] bridge_softmix.c: Recorder/ARI-0000001c;2: Not in SFU mode [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology start [Aug 18 10:34:07] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology destructor [Aug 18 10:34:07] DEBUG[20620] stasis.c: Destroying topic. name: bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66, detail: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20620] stasis.c: Topic 'bridge:6f6fd705-00c9-4b9f-a75f-d7c933b85e66': 0x7f0c3803cb70 destroyed [Aug 18 10:34:07] DEBUG[14146] res_rtp_asterisk.c: (0x7f0c38043ba0) RTP changing ssrc from 2063367591 to 1817792348 due to a source change [Aug 18 10:34:07] DEBUG[13286] stasis/app.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' is 2 interested in calls_0 [Aug 18 10:34:07] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:07] DEBUG[14155] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14153] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[14153] http.c: HTTP Request URI is /ari/channels/robot_213012 [Aug 18 10:34:07] DEBUG[14131] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:07] DEBUG[14131] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14153] http.c: match request [ari/channels/robot_213012] with handler [httpstatus] len 10 [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6550ms with no response [Aug 18 10:34:07] DEBUG[13945] res_stasis_recording.c: 1629282843.237: Sending record(212991_ThJucWjnLOCIgXIZwxMKjVMLdekVScRR.wav) command [Aug 18 10:34:07] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14156] app.c: play_and_record: , /var/spool/asterisk/recording/212991_ThJucWjnLOCIgXIZwxMKjVMLdekVScRR, 'wav' [Aug 18 10:34:07] DEBUG[14156] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:07] VERBOSE[14156] app.c: x=0, open writing: /var/spool/asterisk/recording/212991_ThJucWjnLOCIgXIZwxMKjVMLdekVScRR format: wav, 0x7f0c24113720 [Aug 18 10:34:07] DEBUG[14153] http.c: match request [ari/channels/robot_213012] with handler [phoneprov] len 9 [Aug 18 10:34:07] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14150] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:07] VERBOSE[14150] bridge_channel.c: Channel Announcer/ARI-0000002a;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Hanging up call 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 From: ;tag=as34e313e7 To: ;tag=as51d2bd1a Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" Content-Length: 0 <-------------> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK667494d4;received=159.65.48.104 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as34e313e7 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as51d2bd1a [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dc16f41" [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 (Checking To) --From tag as34e313e7 --To-tag as51d2bd1a [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Stopping retransmission on '59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116993@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c Max-Forwards: 70 From: ;tag=as34e313e7 To: Contact: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (2) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (2) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (4) INVITE - 5 [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:07] DEBUG[14150] bridge.c: Chose bridge technology softmix [Aug 18 10:34:07] VERBOSE[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: switching from simple_bridge technology to softmix [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology constructor [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c9809c220(SIP/zvonobot-0000000e) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[13945] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:07] DEBUG[13945] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:07] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13900] res_stasis_playback.c: 1629282843.234: Sending play(sound:silence/2) command [Aug 18 10:34:07] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) to dummy bridge temporarily [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is leaving simple_bridge technology (dummy) [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology stop [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0ca403e270(Announcer/ARI-0000002a;2) is joining softmix technology [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Announcer/ARI-0000002a;2: [Aug 18 10:34:07] DEBUG[14150] channel.c: Channel Announcer/ARI-0000002a;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Announcer/ARI-0000002a;2: [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Announcer/ARI-0000002a;2: Not in SFU mode [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is joining softmix technology [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: SIP/zvonobot-0000000e: [Aug 18 10:34:07] DEBUG[14154] bridge_softmix.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: starting mixing thread [Aug 18 10:34:07] DEBUG[13900] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:07] DEBUG[13900] http.c: HTTP closing session. Top level [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6567ms with no response [Aug 18 10:34:07] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: SIP/zvonobot-0000000e: [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: SIP/zvonobot-0000000e: Not in SFU mode [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining softmix technology [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Recorder/ARI-00000019;2: [Aug 18 10:34:07] DEBUG[13782] channel.c: Channel 0x7f0c4006f090 'SIP/zvonobot-0000005c' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[14153] http.c: match request [ari/channels/robot_213012] with handler [ari] len 3 [Aug 18 10:34:07] DEBUG[14150] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:07] DEBUG[14150] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Hanging up call 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[14054] res_stasis.c: calls_0: Subscribing to 213084 [Aug 18 10:34:07] DEBUG[14054] stasis/app.c: Channel '213084' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Recorder/ARI-00000019;2: [Aug 18 10:34:07] DEBUG[14150] bridge_softmix.c: Recorder/ARI-00000019;2: Not in SFU mode [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology start [Aug 18 10:34:07] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology destructor [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (2) INVITE - 5 [Aug 18 10:34:07] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14063] res_stasis.c: calls_0: Subscribing to 213085 [Aug 18 10:34:07] DEBUG[14063] stasis/app.c: Channel '213085' is 1 interested in calls_0 [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005c - start 1629282841.120449 answer 0.000000 end 1629282847.928704 dur 6.808 bill 1629282847.928 dispo NO ANSWER [Aug 18 10:34:07] DEBUG[14158] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: starting mixing thread [Aug 18 10:34:07] DEBUG[14160] chan_sip.c: Outgoing Call for 79821116956 [Aug 18 10:34:07] DEBUG[14155] http.c: HTTP Request URI is /ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/record?name=213022_BAYJOCMWmTVQynBoGfsjriMFoVVShhiT&format=wav [Aug 18 10:34:07] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:07] DEBUG[13884] res_stasis_playback.c: 1629282846.255: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (1) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116968@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345 Max-Forwards: 70 From: ;tag=as3a3fa466 To: Contact: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 325427356 325427356 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6386ms with no response [Aug 18 10:34:07] WARNING[20585] chan_sip.c: Hanging up call 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (5) INVITE - 5 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116995@178.62.121.41", nonce="1b522bbc", response="da0414121e3273d111d5c3f47bc85101" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 260407135 260407136 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13804 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP ooh, format changed from none to ulaw [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag [Aug 18 10:34:07] DEBUG[13793] channel.c: Channel 0x7f0c38056500 'SIP/zvonobot-0000005d' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (2) INVITE - 5 [Aug 18 10:34:07] DEBUG[14054] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:07] DEBUG[14054] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[14063] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:07] DEBUG[14063] http.c: HTTP closing session. Top level [Aug 18 10:34:07] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 18904, ms is 2383 [Aug 18 10:34:07] DEBUG[13797] channel.c: Channel 0x7f0c340fb900 'SIP/zvonobot-0000005e' hanging up. Refs: 2 [Aug 18 10:34:07] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[13884] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:07] DEBUG[13884] http.c: HTTP closing session. Top level [Aug 18 10:34:07] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:07] DEBUG[14160] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:07] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005d - start 1629282841.184468 answer 0.000000 end 1629282847.950888 dur 6.766 bill 1629282847.950 dispo NO ANSWER [Aug 18 10:34:07] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:07] DEBUG[14157] http.c: HTTP opening session. Top level [Aug 18 10:34:07] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:07] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:07] DEBUG[14157] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:07] DEBUG[14155] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/record] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14155] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/record] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14073] channel.c: Channel 0x7f0ca0104fa0 'SIP/zvonobot-0000007a' allocated [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:08] DEBUG[14073] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:07] DEBUG[14153] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14155] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/record] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:07] DEBUG[14161] chan_sip.c: Outgoing Call for 79821116955 [Aug 18 10:34:08] DEBUG[14155] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14157] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14162] channel.c: Channel Announcer/ARI-0000002a;1 setting write format path: gsm -> slin [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005e - start 1629282841.495078 answer 0.000000 end 1629282847.963955 dur 6.468 bill 1629282847.963 dispo NO ANSWER [Aug 18 10:34:08] VERBOSE[14160] chan_sip.c: Audio is at 10010 [Aug 18 10:34:08] DEBUG[14157] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Finding handler for channels/robot_213012 [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Finding handler for bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/record [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:08] DEBUG[14159] channel.c: Channel Announcer/ARI-00000028;1 setting write format path: gsm -> slin [Aug 18 10:34:08] DEBUG[14073] res_stasis.c: calls_0: Subscribing to 213088 [Aug 18 10:34:08] DEBUG[14073] stasis/app.c: Channel '213088' is 1 interested in calls_0 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Outgoing Call for 79821116952 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:08] VERBOSE[14165] chan_sip.c: Audio is at 15512 [Aug 18 10:34:08] VERBOSE[14165] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] VERBOSE[14165] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] VERBOSE[14165] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[14162] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:08] VERBOSE[14162] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] DEBUG[14164] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Finding handler for bridges [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:08] VERBOSE[14161] chan_sip.c: Audio is at 14468 [Aug 18 10:34:08] VERBOSE[14161] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] VERBOSE[14161] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] DEBUG[14164] http.c: HTTP Request URI is /ari/channels/213094?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116946&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14164] http.c: match request [ari/channels/213094] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 9 instead [Aug 18 10:34:08] VERBOSE[14161] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14157] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14157] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:08] VERBOSE[14160] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14167] http.c: HTTP opening session. Top level [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1307958254 1307958254 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14926 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3d26e887 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 864, ms is 74 [Aug 18 10:34:08] DEBUG[14159] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:08] DEBUG[14164] http.c: match request [ari/channels/213094] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14073] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:08] DEBUG[14073] http.c: HTTP closing session. Top level [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Finding handler for bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Finding handler for bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:08] DEBUG[14157] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:08] DEBUG[14157] stasis.c: Creating topic. name: bridge:7182caa2-2514-4ffa-b2c8-5bbdd18d9794, detail: [Aug 18 10:34:08] DEBUG[14157] stasis.c: Topic 'bridge:7182caa2-2514-4ffa-b2c8-5bbdd18d9794': 0x7f0c2000ed80 created [Aug 18 10:34:08] DEBUG[14157] stasis.c: Creating topic. name: cache:319/bridge:7182caa2-2514-4ffa-b2c8-5bbdd18d9794, detail: [Aug 18 10:34:08] DEBUG[14157] stasis.c: Topic 'cache:319/bridge:7182caa2-2514-4ffa-b2c8-5bbdd18d9794': 0x7f0c20037230 created [Aug 18 10:34:08] DEBUG[14157] bridge_native_rtp.c: Bridge '7182caa2-2514-4ffa-b2c8-5bbdd18d9794' can not use native RTP bridge as two channels are required [Aug 18 10:34:08] DEBUG[14157] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:08] DEBUG[14157] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:08] DEBUG[14157] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:08] DEBUG[14157] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:08] DEBUG[14157] bridge.c: Bridge 7182caa2-2514-4ffa-b2c8-5bbdd18d9794: calling simple_bridge technology constructor [Aug 18 10:34:08] DEBUG[14157] bridge.c: Bridge 7182caa2-2514-4ffa-b2c8-5bbdd18d9794: calling simple_bridge technology start [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Finding handler for robot_213012 [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking channels create: Didn't match robot_213012 [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14153] res_ari.c: Checking channels externalMedia: Didn't match robot_213012 [Aug 18 10:34:08] DEBUG[14153] res_ari.c: No explicit handler found for robot_213012. Using wildcard channelId. [Aug 18 10:34:08] VERBOSE[14159] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Initializing initreq for method INVITE - callid 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116952@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 3 [ 52]: From: ;tag=as64e6e544 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 6 [ 60]: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] VERBOSE[14165] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #106 [Aug 18 10:34:08] DEBUG[14165] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14169] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Finding handler for fbfe71c6-df7c-4b8c-8b66-37207aaf9846 [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14155] res_ari.c: No explicit handler found for fbfe71c6-df7c-4b8c-8b66-37207aaf9846. Using wildcard bridgeId. [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Finding handler for record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:08] DEBUG[14155] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:08] DEBUG[14155] stasis.c: Creating topic. name: channel:1629282848.273, detail: [Aug 18 10:34:08] DEBUG[14155] stasis.c: Topic 'channel:1629282848.273': 0x7f0c180bacb0 created [Aug 18 10:34:08] DEBUG[14155] stasis.c: Creating topic. name: cache:320/channel:1629282848.273, detail: [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14169] http.c: HTTP Request URI is /ari/channels/212991/snoop?app=calls_0&spy=in [Aug 18 10:34:08] DEBUG[14170] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14155] stasis.c: Topic 'cache:320/channel:1629282848.273': 0x7f0c180bfd90 created [Aug 18 10:34:08] DEBUG[14164] http.c: match request [ari/channels/213094] with handler [ari] len 3 [Aug 18 10:34:08] VERBOSE[14165] dial.c: Called zvonobot/79821116952 [Aug 18 10:34:08] VERBOSE[14160] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Initializing initreq for method INVITE - callid 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116955@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 3 [ 52]: From: ;tag=as40bb47c8 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 6 [ 60]: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] VERBOSE[14161] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Aug 18 10:34:08] DEBUG[14161] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14169] http.c: match request [ari/channels/212991/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14169] http.c: match request [ari/channels/212991/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14169] http.c: match request [ari/channels/212991/snoop] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14169] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Finding handler for channels/212991/snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Finding handler for 212991 [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channels create: Didn't match 212991 [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channels externalMedia: Didn't match 212991 [Aug 18 10:34:08] DEBUG[14169] res_ari.c: No explicit handler found for 212991. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Finding handler for snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:08] DEBUG[14169] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:08] DEBUG[14170] http.c: HTTP Request URI is /ari/channels/213096?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116944&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14170] http.c: match request [ari/channels/213096] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14170] http.c: match request [ari/channels/213096] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14170] http.c: match request [ari/channels/213096] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14170] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14170] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Finding handler for channels/213096 [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Finding handler for 213096 [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking channels create: Didn't match 213096 [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14170] res_ari.c: Checking channels externalMedia: Didn't match 213096 [Aug 18 10:34:08] DEBUG[14170] res_ari.c: No explicit handler found for 213096. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1307958254 1307958254 IN IP4 178.62.121.41 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14926 RTP/AVP 0 8 101 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 (Checking To) --From tag as7d114a77 --To-tag as3d26e887 [Aug 18 10:34:08] DEBUG[14157] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:08] DEBUG[14157] http.c: HTTP closing session. Top level [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1120, ms is 90 [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14160] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14167] http.c: HTTP Request URI is /ari/channels/213095?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116945&callerId=74950493843 [Aug 18 10:34:08] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14176] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Stopping retransmission on '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14167] http.c: match request [ari/channels/213095] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14164] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14176] http.c: HTTP Request URI is /ari/channels/213097?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116943&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14176] http.c: match request [ari/channels/213097] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14176] http.c: match request [ari/channels/213097] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14176] http.c: match request [ari/channels/213097] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14176] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14176] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Finding handler for channels/213097 [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Finding handler for 213097 [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking channels create: Didn't match 213097 [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14176] res_ari.c: Checking channels externalMedia: Didn't match 213097 [Aug 18 10:34:08] DEBUG[14176] res_ari.c: No explicit handler found for 213097. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 960, ms is 80 [Aug 18 10:34:08] DEBUG[14167] http.c: match request [ari/channels/213095] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Strict routing enforced for session 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14161] dial.c: Called zvonobot/79821116955 [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:08] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:08] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117075@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK74e7da8b Max-Forwards: 70 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Initializing initreq for method INVITE - callid 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116956@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 3 [ 52]: From: ;tag=as4d13c830 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 6 [ 60]: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14167] http.c: match request [ari/channels/213095] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14167] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (5) INVITE - 5 [Aug 18 10:34:08] DEBUG[14180] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14167] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Finding handler for channels/213095 [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Finding handler for 213095 [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking channels create: Didn't match 213095 [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14167] res_ari.c: Checking channels externalMedia: Didn't match 213095 [Aug 18 10:34:08] DEBUG[14167] res_ari.c: No explicit handler found for 213095. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116990@178.62.121.41", nonce="068a7741", response="119167baf609206fb4c8e9c8920f6f8c" Date: Wed, 18 Aug 2021 10:34:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712771033 1712771034 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10086 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6440ms with no response [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[13801] channel.c: Channel 0x7f0c78050c90 'SIP/zvonobot-0000005f' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[14164] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Finding handler for channels/213094 [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Finding handler for 213094 [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking channels create: Didn't match 213094 [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14164] res_ari.c: Checking channels externalMedia: Didn't match 213094 [Aug 18 10:34:08] DEBUG[14164] res_ari.c: No explicit handler found for 213094. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000005f - start 1629282841.591506 answer 0.000000 end 1629282848.183401 dur 6.591 bill 1629282848.183 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[14182] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14182] http.c: HTTP Request URI is /ari/channels/213098?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116942&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14182] http.c: match request [ari/channels/213098] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14182] http.c: match request [ari/channels/213098] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14182] http.c: match request [ari/channels/213098] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14182] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14182] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Finding handler for channels/213098 [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Finding handler for 213098 [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking channels create: Didn't match 213098 [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14182] res_ari.c: Checking channels externalMedia: Didn't match 213098 [Aug 18 10:34:08] DEBUG[14182] res_ari.c: No explicit handler found for 213098. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14183] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: Allocating new SIP dialog for 2577af086aa9f70d0e42e4455d8ac09b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14170] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c018b60' [Aug 18 10:34:08] DEBUG[14180] http.c: HTTP Request URI is /ari/channels/213099?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116941&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) RTP allocated port 14552 [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14180] http.c: match request [ari/channels/213099] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14180] http.c: match request [ari/channels/213099] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14180] http.c: match request [ari/channels/213099] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE creating session 0.0.0.0:14552 (14552) [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE create [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE add system candidates [Aug 18 10:34:08] DEBUG[14170] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[14170] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE add candidate: 159.65.48.104:14552, 2130706431 [Aug 18 10:34:08] DEBUG[14170] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14170] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE add candidate: 10.131.0.10:14552, 2130706431 [Aug 18 10:34:08] DEBUG[14170] rtp_engine.c: RTP instance '0x7f0c7c018b60' is setup and ready to go [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE stopped [Aug 18 10:34:08] DEBUG[14170] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14170] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[14170] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[14170] res_rtp_asterisk.c: (0x7f0c7c018b60) RTCP setup on RTP instance [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #111 (4) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #111)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as0a75a671 Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" Content-Length: 0 <-------------> [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e988fac;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a75a671 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[13056] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:34:08] DEBUG[14180] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6309d33d" [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14170] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14185] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[14184] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14183] http.c: HTTP Request URI is /ari/channels/213103?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116937&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14187] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14183] http.c: match request [ari/channels/213103] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] DEBUG[14170] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14170] chan_sip.c: SIP call-id changed from '2577af086aa9f70d0e42e4455d8ac09b@127.0.1.1:5060' to '21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14170] stasis.c: Creating topic. name: channel:213096, detail: [Aug 18 10:34:08] DEBUG[14170] stasis.c: Topic 'channel:213096': 0x7f0c7c015c40 created [Aug 18 10:34:08] DEBUG[14170] stasis.c: Creating topic. name: cache:321/channel:213096, detail: [Aug 18 10:34:08] DEBUG[14170] stasis.c: Topic 'cache:321/channel:213096': 0x7f0c7c0c6230 created [Aug 18 10:34:08] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 848, ms is 73 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] DEBUG[14187] http.c: HTTP Request URI is /ari/channels/213101?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116939&callerId=74950493843 [Aug 18 10:34:08] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP got report of 100 bytes from 178.62.121.41:10225 [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14185] http.c: HTTP Request URI is /ari/channels/213102?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116938&callerId=74950493843 [Aug 18 10:34:08] DEBUG[14183] http.c: match request [ari/channels/213103] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14180] http.c: HTTP consuming request body [Aug 18 10:34:08] VERBOSE[14160] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Finding handler for channels/213099 [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Finding handler for 213099 [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking channels create: Didn't match 213099 [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14180] res_ari.c: Checking channels externalMedia: Didn't match 213099 [Aug 18 10:34:08] DEBUG[14180] res_ari.c: No explicit handler found for 213099. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[14184] http.c: HTTP Request URI is /ari/channels/213100?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116940&callerId=74950493843 [Aug 18 10:34:08] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 720, ms is 65 [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14187] http.c: match request [ari/channels/213101] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14184] http.c: match request [ari/channels/213100] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14183] http.c: match request [ari/channels/213103] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14183] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #68 [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14185] http.c: match request [ari/channels/213102] with handler [httpstatus] len 10 [Aug 18 10:34:08] DEBUG[14183] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14185] http.c: match request [ari/channels/213102] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14160] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14184] http.c: match request [ari/channels/213100] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14185] http.c: match request [ari/channels/213102] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as0a75a671 [Aug 18 10:34:08] DEBUG[14187] http.c: match request [ari/channels/213101] with handler [phoneprov] len 9 [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Finding handler for channels/213103 [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14187] http.c: match request [ari/channels/213101] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[14187] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14184] http.c: match request [ari/channels/213100] with handler [ari] len 3 [Aug 18 10:34:08] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] VERBOSE[14160] dial.c: Called zvonobot/79821116956 [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Finding handler for 213103 [Aug 18 10:34:08] DEBUG[14187] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14184] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14185] http.c: Match made with [ari] [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking channels create: Didn't match 213103 [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Stopping retransmission on '16541045053beef31ed8e5361337aa22@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14183] res_ari.c: Checking channels externalMedia: Didn't match 213103 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14183] res_ari.c: No explicit handler found for 213103. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Finding handler for channels/213101 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116989@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939 Max-Forwards: 70 From: ;tag=as79336d5f To: Contact: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:08] DEBUG[14059] channel.c: Channel 0x7f0ca80ecaf0 'SIP/zvonobot-0000007b' allocated [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:08] DEBUG[14059] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14184] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14185] http.c: HTTP consuming request body [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14059] res_stasis.c: calls_0: Subscribing to 213086 [Aug 18 10:34:08] DEBUG[14059] stasis/app.c: Channel '213086' is 1 interested in calls_0 [Aug 18 10:34:08] DEBUG[14059] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:08] DEBUG[14059] http.c: HTTP closing session. Top level [Aug 18 10:34:08] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: Allocating new SIP dialog for 60de19496f3481f848a9fd9a4fd80e2f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14182] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c80062990' [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) RTP allocated port 15826 [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE creating session 0.0.0.0:15826 (15826) [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE create [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE add system candidates [Aug 18 10:34:08] DEBUG[14182] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[14182] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE add candidate: 159.65.48.104:15826, 2130706431 [Aug 18 10:34:08] DEBUG[14182] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14182] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE add candidate: 10.131.0.10:15826, 2130706431 [Aug 18 10:34:08] DEBUG[14182] rtp_engine.c: RTP instance '0x7f0c80062990' is setup and ready to go [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) ICE stopped [Aug 18 10:34:08] DEBUG[14182] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14182] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[14182] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[14182] res_rtp_asterisk.c: (0x7f0c80062990) RTCP setup on RTP instance [Aug 18 10:34:08] VERBOSE[14182] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] DEBUG[14182] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14182] chan_sip.c: SIP call-id changed from '60de19496f3481f848a9fd9a4fd80e2f@127.0.1.1:5060' to '1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14182] stasis.c: Creating topic. name: channel:213098, detail: [Aug 18 10:34:08] DEBUG[14182] stasis.c: Topic 'channel:213098': 0x7f0c8006ab30 created [Aug 18 10:34:08] DEBUG[14182] stasis.c: Creating topic. name: cache:322/channel:213098, detail: [Aug 18 10:34:08] DEBUG[14182] stasis.c: Topic 'cache:322/channel:213098': 0x7f0c8006b5b0 created [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Finding handler for channels/213102 [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Finding handler for 213102 [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking channels create: Didn't match 213102 [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14185] res_ari.c: Checking channels externalMedia: Didn't match 213102 [Aug 18 10:34:08] DEBUG[14185] res_ari.c: No explicit handler found for 213102. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Outgoing Call for 79821116954 [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Finding handler for channels/213100 [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13077] res_rtp_asterisk.c: (0x7f0c18009d50) RTP audio difference is 688, ms is 63 [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: Allocating new SIP dialog for 351ffb3e197514b06d5dfedf39fa10f3@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14176] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c78040090' [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) RTP allocated port 18438 [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE creating session 0.0.0.0:18438 (18438) [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE create [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE add system candidates [Aug 18 10:34:08] DEBUG[14176] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (1) INVITE - 5 [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 8 instead [Aug 18 10:34:08] DEBUG[14176] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[13870] channel.c: Channel 0x7f0c900818f0 'Announcer/ARI-0000002b;1' allocated [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Finding handler for channels [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:08] DEBUG[13870] stasis.c: Creating topic. name: channel:1629282848.276, detail: [Aug 18 10:34:08] DEBUG[13870] stasis.c: Topic 'channel:1629282848.276': 0x7f0c900288b0 created [Aug 18 10:34:08] DEBUG[13870] stasis.c: Creating topic. name: cache:323/channel:1629282848.276, detail: [Aug 18 10:34:08] DEBUG[13870] stasis.c: Topic 'cache:323/channel:1629282848.276': 0x7f0c900268e0 created [Aug 18 10:34:08] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14074] channel.c: Channel 0x7f0ca4057750 'SIP/zvonobot-0000007c' allocated [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:08] DEBUG[14074] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:08] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Finding handler for 213101 [Aug 18 10:34:08] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 952, ms is 139 [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE add candidate: 159.65.48.104:18438, 2130706431 [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking channels create: Didn't match 213101 [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:08] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: Allocating new SIP dialog for 6de1c6d82292f485561d4cb4481ec0da@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14167] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c74063020' [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) RTP allocated port 14556 [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE creating session 0.0.0.0:14556 (14556) [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE create [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE add system candidates [Aug 18 10:34:08] DEBUG[14167] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[14167] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE add candidate: 159.65.48.104:14556, 2130706431 [Aug 18 10:34:08] DEBUG[14167] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14167] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE add candidate: 10.131.0.10:14556, 2130706431 [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14189] chan_sip.c: Audio is at 10836 [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (1) INVITE - 5 [Aug 18 10:34:08] DEBUG[14167] rtp_engine.c: RTP instance '0x7f0c74063020' is setup and ready to go [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6604ms with no response [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6584ms with no response [Aug 18 10:34:08] DEBUG[13810] channel.c: Channel 0x7f0c9007e3b0 'SIP/zvonobot-00000060' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000060 - start 1629282841.770412 answer 0.000000 end 1629282848.528200 dur 6.757 bill 1629282848.528 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:08] DEBUG[14187] res_ari.c: Checking channels externalMedia: Didn't match 213101 [Aug 18 10:34:08] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14187] res_ari.c: No explicit handler found for 213101. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[14176] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Finding handler for 213100 [Aug 18 10:34:08] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 688, ms is 63 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) ICE stopped [Aug 18 10:34:08] DEBUG[14167] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14167] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[14176] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] VERBOSE[14189] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking channels create: Didn't match 213100 [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14184] res_ari.c: Checking channels externalMedia: Didn't match 213100 [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: Allocating new SIP dialog for 4458835f62cfe46c53e3904e5f12bbdb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14164] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3808f7a0' [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) RTP allocated port 19712 [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE creating session 0.0.0.0:19712 (19712) [Aug 18 10:34:08] DEBUG[14184] res_ari.c: No explicit handler found for 213100. Using wildcard channelId. [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13811] channel.c: Channel 0x7f0c841031a0 'SIP/zvonobot-00000062' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000062 - start 1629282841.819951 answer 0.000000 end 1629282848.582717 dur 6.762 bill 1629282848.582 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[14167] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] VERBOSE[14189] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] VERBOSE[14189] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Initializing initreq for method INVITE - callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116954@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 3 [ 52]: From: ;tag=as2d7c4d21 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 6 [ 60]: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6603ms with no response [Aug 18 10:34:08] DEBUG[13541] res_rtp_asterisk.c: (0x7f0c080871b0) RTP audio difference is 720, ms is 65 [Aug 18 10:34:08] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE create [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6566ms with no response [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[13636] channel.c: Channel 0x7f0c10106720 'SIP/zvonobot-00000051' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000051 - start 1629282838.729099 answer 0.000000 end 1629282848.628404 dur 9.899 bill 1629282848.628 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[13631] channel.c: Channel 0x7f0cb40628c0 'SIP/zvonobot-00000050' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000050 - start 1629282838.655524 answer 0.000000 end 1629282848.630694 dur 9.975 bill 1629282848.630 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE add system candidates [Aug 18 10:34:08] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE add candidate: 10.131.0.10:18438, 2130706431 [Aug 18 10:34:08] DEBUG[14176] rtp_engine.c: RTP instance '0x7f0c78040090' is setup and ready to go [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) ICE stopped [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (1) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14167] res_rtp_asterisk.c: (0x7f0c74063020) RTCP setup on RTP instance [Aug 18 10:34:08] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1168, ms is 93 [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14176] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[14164] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] VERBOSE[14167] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6553ms with no response [Aug 18 10:34:08] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14164] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14176] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[14176] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[14176] res_rtp_asterisk.c: (0x7f0c78040090) RTCP setup on RTP instance [Aug 18 10:34:08] VERBOSE[14176] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] DEBUG[14176] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14176] chan_sip.c: SIP call-id changed from '351ffb3e197514b06d5dfedf39fa10f3@127.0.1.1:5060' to '4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14176] stasis.c: Creating topic. name: channel:213097, detail: [Aug 18 10:34:08] DEBUG[14176] stasis.c: Topic 'channel:213097': 0x7f0c780a4760 created [Aug 18 10:34:08] DEBUG[14176] stasis.c: Creating topic. name: cache:324/channel:213097, detail: [Aug 18 10:34:08] DEBUG[14176] stasis.c: Topic 'cache:324/channel:213097': 0x7f0c780a51e0 created [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] VERBOSE[14189] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #26 [Aug 18 10:34:08] DEBUG[14189] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[13819] channel.c: Channel 0x7f0c8809bb30 'SIP/zvonobot-00000061' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000061 - start 1629282842.005888 answer 0.000000 end 1629282848.681216 dur 6.675 bill 1629282848.681 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14189] dial.c: Called zvonobot/79821116954 [Aug 18 10:34:08] DEBUG[14167] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14069] channel.c: Channel 0x7f0c98081820 'SIP/zvonobot-0000007d' allocated [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:08] DEBUG[14069] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[13151] channel.c: Channel 0x7f0c2001ad30 'Announcer/ARI-00000002;2' destroying [Aug 18 10:34:08] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE add candidate: 159.65.48.104:19712, 2130706431 [Aug 18 10:34:08] DEBUG[13151] stasis.c: Destroying topic. name: cache:58/channel:1629282829.49, detail: [Aug 18 10:34:08] DEBUG[13151] stasis.c: Topic 'cache:58/channel:1629282829.49': 0x7f0c2000cbc0 destroyed [Aug 18 10:34:08] DEBUG[13151] stasis.c: Destroying topic. name: channel:1629282829.49, detail: [Aug 18 10:34:08] DEBUG[13151] stasis.c: Topic 'channel:1629282829.49': 0x7f0c2000d150 destroyed [Aug 18 10:34:08] DEBUG[14074] res_stasis.c: calls_0: Subscribing to 213089 [Aug 18 10:34:08] DEBUG[14074] stasis/app.c: Channel '213089' is 1 interested in calls_0 [Aug 18 10:34:08] DEBUG[14074] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:08] DEBUG[14069] res_stasis.c: calls_0: Subscribing to 213087 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:08] DEBUG[14069] stasis/app.c: Channel '213087' is 1 interested in calls_0 [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call leg/transaction does not exist [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3725d74e [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as3725d74e [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 474 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6578ms with no response [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (5) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116970@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838 Max-Forwards: 70 From: ;tag=as2eb39fa6 To: Contact: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 468185486 468185486 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14444 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (2) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Outgoing Call for 79821116951 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:08] VERBOSE[14190] chan_sip.c: Audio is at 18668 [Aug 18 10:34:08] VERBOSE[14190] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] VERBOSE[14190] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] VERBOSE[14190] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Initializing initreq for method INVITE - callid 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116951@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 3 [ 52]: From: ;tag=as2ed109a6 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 6 [ 60]: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14069] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:08] DEBUG[14069] http.c: HTTP closing session. Top level [Aug 18 10:34:08] DEBUG[14074] http.c: HTTP closing session. Top level [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Outgoing Call for 79821116953 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] VERBOSE[14190] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #105 [Aug 18 10:34:08] DEBUG[14190] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (2) INVITE - 5 [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[13820] channel.c: Channel 0x7f0c8c0f9790 'SIP/zvonobot-00000063' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000063 - start 1629282842.029013 answer 0.000000 end 1629282848.744267 dur 6.715 bill 1629282848.744 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14167] chan_sip.c: SIP call-id changed from '6de1c6d82292f485561d4cb4481ec0da@127.0.1.1:5060' to '2d4029193f64fb721f43803f29facceb@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14167] stasis.c: Creating topic. name: channel:213095, detail: [Aug 18 10:34:08] DEBUG[14167] stasis.c: Topic 'channel:213095': 0x7f0c74053a80 created [Aug 18 10:34:08] DEBUG[14167] stasis.c: Creating topic. name: cache:325/channel:213095, detail: [Aug 18 10:34:08] DEBUG[14167] stasis.c: Topic 'cache:325/channel:213095': 0x7f0c740993c0 created [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTCP got report of 100 bytes from 178.62.121.41:12913 [Aug 18 10:34:08] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE add candidate: 10.131.0.10:19712, 2130706431 [Aug 18 10:34:08] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: Allocating new SIP dialog for 4ce083d336279a674acf0f5777e969ff@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14183] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c03ed40' [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) RTP allocated port 10618 [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE creating session 0.0.0.0:10618 (10618) [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE create [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE add system candidates [Aug 18 10:34:08] DEBUG[14183] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[14183] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE add candidate: 159.65.48.104:10618, 2130706431 [Aug 18 10:34:08] DEBUG[14183] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14183] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE add candidate: 10.131.0.10:10618, 2130706431 [Aug 18 10:34:08] DEBUG[14183] rtp_engine.c: RTP instance '0x7f0c8c03ed40' is setup and ready to go [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE stopped [Aug 18 10:34:08] DEBUG[14183] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14183] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[14183] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[14183] res_rtp_asterisk.c: (0x7f0c8c03ed40) RTCP setup on RTP instance [Aug 18 10:34:08] VERBOSE[14183] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] DEBUG[14183] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14183] chan_sip.c: SIP call-id changed from '4ce083d336279a674acf0f5777e969ff@127.0.1.1:5060' to '1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:08] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 656, ms is 102 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (5) BYE - 8 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:08] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:08] DEBUG[14164] rtp_engine.c: RTP instance '0x7f0c3808f7a0' is setup and ready to go [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) ICE stopped [Aug 18 10:34:08] DEBUG[14164] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:08] DEBUG[14164] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:08] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTCP got report of 100 bytes from 178.62.121.41:16139 [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:08] VERBOSE[14190] dial.c: Called zvonobot/79821116951 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: BYE sip:79821117041@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39b3ab90 Max-Forwards: 70 From: ;tag=as3a1fc7ed To: ;tag=as3f55a57d Call-ID: 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117041@178.62.121.41:5060", nonce="22eff029", response="64a149512afc8a5ee06b37f83da9b30e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (2) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:08] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] VERBOSE[14193] chan_sip.c: Audio is at 18262 [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 848, ms is 73 [Aug 18 10:34:08] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[14183] stasis.c: Creating topic. name: channel:213103, detail: [Aug 18 10:34:08] DEBUG[14164] res_rtp_asterisk.c: (0x7f0c3808f7a0) RTCP setup on RTP instance [Aug 18 10:34:08] VERBOSE[14193] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[14193] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 From: ;tag=as63ca65c0 To: ;tag=as56bda81b Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" Content-Length: 0 <-------------> [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bb0e0f1;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56bda81b [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b12677" [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag as56bda81b [Aug 18 10:34:08] DEBUG[14183] stasis.c: Topic 'channel:213103': 0x7f0c8c07a110 created [Aug 18 10:34:08] DEBUG[14183] stasis.c: Creating topic. name: cache:326/channel:213103, detail: [Aug 18 10:34:08] DEBUG[14183] stasis.c: Topic 'cache:326/channel:213103': 0x7f0c8c04f7e0 created [Aug 18 10:34:08] VERBOSE[14164] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Stopping retransmission on '63c7430f0c6f8039194559375e330124@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:08] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116987@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5 Max-Forwards: 70 From: ;tag=as63ca65c0 To: Contact: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] VERBOSE[14193] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #26 (1) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #26)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #105 (1) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #105)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (5) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 221776054 221776054 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10694 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d218141 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6bcc6b47 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 221776054 221776054 IN IP4 178.62.121.41 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10694 RTP/AVP 0 8 101 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 (Checking To) --From tag as2d218141 --To-tag as6bcc6b47 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Stopping retransmission on '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:08] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Strict routing enforced for session 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:08] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:08] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Initializing initreq for method INVITE - callid 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117025@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK28edfc48 Max-Forwards: 70 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (4) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (5) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[14164] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116953@178.62.121.41 SIP/2.0 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:08] DEBUG[14164] chan_sip.c: SIP call-id changed from '4458835f62cfe46c53e3904e5f12bbdb@127.0.1.1:5060' to '0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060' [Aug 18 10:34:08] DEBUG[14164] stasis.c: Creating topic. name: channel:213094, detail: [Aug 18 10:34:08] DEBUG[14164] stasis.c: Topic 'channel:213094': 0x7f0c3804c510 created [Aug 18 10:34:08] DEBUG[14164] stasis.c: Creating topic. name: cache:327/channel:213094, detail: [Aug 18 10:34:08] DEBUG[14164] stasis.c: Topic 'cache:327/channel:213094': 0x7f0c3805e1b0 created [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call leg/transaction does not exist [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3725d74e [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 3 [ 52]: From: ;tag=as42198afd [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:08] DEBUG[14185] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14185] chan_sip.c: Allocating new SIP dialog for 791bcf677c35c7be33e64b365a89e7fb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:08] DEBUG[14185] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c940b4340' [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) RTP allocated port 14986 [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE creating session 0.0.0.0:14986 (14986) [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE create [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE add system candidates [Aug 18 10:34:08] DEBUG[14185] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:08] DEBUG[14185] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE add candidate: 159.65.48.104:14986, 2130706431 [Aug 18 10:34:08] DEBUG[14185] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:08] DEBUG[14185] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE add candidate: 10.131.0.10:14986, 2130706431 [Aug 18 10:34:08] DEBUG[14185] rtp_engine.c: RTP instance '0x7f0c940b4340' is setup and ready to go [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 6 [ 60]: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:08] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:08] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:08] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as3725d74e [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 474 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (4) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (3) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (4) INVITE - 5 [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:08] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6126ms with no response [Aug 18 10:34:08] WARNING[20585] chan_sip.c: Hanging up call 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:08] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:08 GMT [Aug 18 10:34:08] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:08] DEBUG[13645] channel.c: Channel 0x2c6e150 'SIP/zvonobot-00000053' hanging up. Refs: 2 [Aug 18 10:34:08] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000053 - start 1629282838.991760 answer 0.000000 end 1629282848.994851 dur 10.003 bill 1629282848.994 dispo NO ANSWER [Aug 18 10:34:08] DEBUG[14193] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:08] DEBUG[14200] http.c: HTTP opening session. Top level [Aug 18 10:34:08] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] VERBOSE[14193] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14193] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #72 [Aug 18 10:34:09] DEBUG[14193] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:08] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) ICE stopped [Aug 18 10:34:08] DEBUG[14200] http.c: HTTP Request URI is /ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: Allocating new SIP dialog for 2a0252e45f4c6586075aa91046271766@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:09] DEBUG[14184] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c88031270' [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) RTP allocated port 11552 [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE creating session 0.0.0.0:11552 (11552) [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE create [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:09] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:09] DEBUG[14200] http.c: match request [ari/bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play] with handler [ari] len 3 [Aug 18 10:34:09] VERBOSE[14193] dial.c: Called zvonobot/79821116953 [Aug 18 10:34:09] DEBUG[14200] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Finding handler for bridges/0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3/play [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Finding handler for bridges [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:09] DEBUG[14185] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:09] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:09] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE add system candidates [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:09] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14185] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:09] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14084] channel.c: Channel 0x7f0cb4073e80 'SIP/zvonobot-0000007f' allocated [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:09] DEBUG[14084] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:09] DEBUG[14084] res_stasis.c: calls_0: Subscribing to 213093 [Aug 18 10:34:09] DEBUG[14084] stasis/app.c: Channel '213093' is 1 interested in calls_0 [Aug 18 10:34:09] DEBUG[14084] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Finding handler for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3 [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14084] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[14200] res_ari.c: No explicit handler found for 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3. Using wildcard bridgeId. [Aug 18 10:34:09] DEBUG[14185] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:09] DEBUG[14185] res_rtp_asterisk.c: (0x7f0c940b4340) RTCP setup on RTP instance [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Finding handler for play [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: Allocating new SIP dialog for 21cd085c00839abe68a9c09563bd2332@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:09] DEBUG[14187] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c900af9f0' [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) RTP allocated port 16540 [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE creating session 0.0.0.0:16540 (16540) [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE create [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Outgoing Call for 79821116947 [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14184] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14200] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 From: ;tag=as6093d024 To: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2cec2e2d;received=159.65.48.104 [Aug 18 10:34:09] VERBOSE[14185] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:09] DEBUG[14185] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14185] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:09] DEBUG[14185] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:09] DEBUG[14185] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14185] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14185] chan_sip.c: SIP call-id changed from '791bcf677c35c7be33e64b365a89e7fb@127.0.1.1:5060' to '2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060' [Aug 18 10:34:09] DEBUG[14185] stasis.c: Creating topic. name: channel:213102, detail: [Aug 18 10:34:09] DEBUG[14185] stasis.c: Topic 'channel:213102': 0x7f0c940ae940 created [Aug 18 10:34:09] DEBUG[14185] stasis.c: Creating topic. name: cache:328/channel:213102, detail: [Aug 18 10:34:09] DEBUG[14185] stasis.c: Topic 'cache:328/channel:213102': 0x7f0c940af6a0 created [Aug 18 10:34:09] DEBUG[14184] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6093d024 [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:09] VERBOSE[14202] chan_sip.c: Audio is at 10588 [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE add candidate: 159.65.48.104:11552, 2130706431 [Aug 18 10:34:09] VERBOSE[14202] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060 (Checking To) --From tag as6093d024 --To-tag [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE add system candidates [Aug 18 10:34:09] DEBUG[14187] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:09] DEBUG[14187] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE add candidate: 159.65.48.104:16540, 2130706431 [Aug 18 10:34:09] DEBUG[14187] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:09] DEBUG[14187] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE add candidate: 10.131.0.10:16540, 2130706431 [Aug 18 10:34:09] DEBUG[14187] rtp_engine.c: RTP instance '0x7f0c900af9f0' is setup and ready to go [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) ICE stopped [Aug 18 10:34:09] DEBUG[14187] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:09] DEBUG[14187] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:09] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[14202] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:09] VERBOSE[14202] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:09] WARNING[13944] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000026;1 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Initializing initreq for method INVITE - callid 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116947@178.62.121.41 SIP/2.0 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 3 [ 52]: From: ;tag=as293a990c [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fdaba5d4e6664dc6f8664aa150c059b@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:09] DEBUG[14187] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 6 [ 60]: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #26 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #26)) [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14184] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #105 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[14187] res_rtp_asterisk.c: (0x7f0c900af9f0) RTCP setup on RTP instance [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #105)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:09 GMT [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (5) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:09] DEBUG[14184] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 815742320 815742320 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17664 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] VERBOSE[14202] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #78 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[14202] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[14187] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:09] DEBUG[14187] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14187] chan_sip.c: SIP call-id changed from '21cd085c00839abe68a9c09563bd2332@127.0.1.1:5060' to '7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (3) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (1) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (4) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (5) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (5) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 From: ;tag=as28933467 To: ;tag=as5b70cf89 Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1807314912 1807314912 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28933467 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5b70cf89 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1807314912 1807314912 IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14088 RTP/AVP 0 8 101 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking To) --From tag as28933467 --To-tag as5b70cf89 [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14187] stasis.c: Creating topic. name: channel:213101, detail: [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE add candidate: 10.131.0.10:11552, 2130706431 [Aug 18 10:34:09] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:09] DEBUG[14184] rtp_engine.c: RTP instance '0x7f0c88031270' is setup and ready to go [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) ICE stopped [Aug 18 10:34:09] DEBUG[14184] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:09] DEBUG[14184] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:09] DEBUG[14184] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:09] VERBOSE[14202] dial.c: Called zvonobot/79821116947 [Aug 18 10:34:09] DEBUG[14184] res_rtp_asterisk.c: (0x7f0c88031270) RTCP setup on RTP instance [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[14184] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:09] DEBUG[14184] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14184] chan_sip.c: SIP call-id changed from '2a0252e45f4c6586075aa91046271766@127.0.1.1:5060' to '3782ef707142714164cf352b663534ff@159.65.48.104:5060' [Aug 18 10:34:09] DEBUG[14184] stasis.c: Creating topic. name: channel:213100, detail: [Aug 18 10:34:09] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:09] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:09] DEBUG[14184] stasis.c: Topic 'channel:213100': 0x7f0c8806f040 created [Aug 18 10:34:09] DEBUG[14184] stasis.c: Creating topic. name: cache:329/channel:213100, detail: [Aug 18 10:34:09] DEBUG[14184] stasis.c: Topic 'cache:329/channel:213100': 0x7f0c880a2be0 created [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: Allocating new SIP dialog for 24e7688d7df856a427c7e0d045dd3abc@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117073@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57dc7406 Max-Forwards: 70 From: ;tag=as28933467 To: ;tag=as5b70cf89 Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[14187] stasis.c: Topic 'channel:213101': 0x7f0c900811a0 created [Aug 18 10:34:09] DEBUG[14187] stasis.c: Creating topic. name: cache:330/channel:213101, detail: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14187] stasis.c: Topic 'cache:330/channel:213101': 0x7f0c900755c0 created [Aug 18 10:34:09] DEBUG[14180] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c840953a0' [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) RTP allocated port 10796 [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE creating session 0.0.0.0:10796 (10796) [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE create [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE add system candidates [Aug 18 10:34:09] DEBUG[14180] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:09] DEBUG[14180] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE add candidate: 159.65.48.104:10796, 2130706431 [Aug 18 10:34:09] DEBUG[14180] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:09] DEBUG[14180] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE add candidate: 10.131.0.10:10796, 2130706431 [Aug 18 10:34:09] DEBUG[14180] rtp_engine.c: RTP instance '0x7f0c840953a0' is setup and ready to go [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) ICE stopped [Aug 18 10:34:09] DEBUG[14180] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:09] DEBUG[14180] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:09] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14085] channel.c: Channel 0x2c47a50 'SIP/zvonobot-0000007e' allocated [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:09] DEBUG[14085] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:09] DEBUG[14085] res_stasis.c: calls_0: Subscribing to 213090 [Aug 18 10:34:09] DEBUG[14085] stasis/app.c: Channel '213090' is 1 interested in calls_0 [Aug 18 10:34:09] DEBUG[14085] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:09] DEBUG[14085] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Outgoing Call for 79821116950 [Aug 18 10:34:09] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 From: ;tag=as1dc5ead1 To: ;tag=as741846ee Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:09] VERBOSE[14204] chan_sip.c: Audio is at 17384 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b67f4a3;received=159.65.48.104 [Aug 18 10:34:09] VERBOSE[14204] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1dc5ead1 [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as741846ee [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14180] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] VERBOSE[14204] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268549f1" [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:09] VERBOSE[14204] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 (Checking To) --From tag as1dc5ead1 --To-tag as741846ee [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Initializing initreq for method INVITE - callid 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116950@178.62.121.41 SIP/2.0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116996@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b5954c2 Max-Forwards: 70 From: ;tag=as1dc5ead1 To: Contact: Call-ID: 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 3 [ 52]: From: ;tag=as05f1fb09 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 6 [ 60]: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:09 GMT [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (1) INVITE - 5 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] VERBOSE[14204] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (5) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #77 [Aug 18 10:34:09] DEBUG[14204] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14180] res_rtp_asterisk.c: (0x7f0c840953a0) RTCP setup on RTP instance [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #26 (3) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #26)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 From: ;tag=as02885f54 To: ;tag=as5580464d Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06142ef0;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as02885f54 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5580464d [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] VERBOSE[14204] dial.c: Called zvonobot/79821116950 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d33e27" [Aug 18 10:34:09] VERBOSE[14180] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:09] DEBUG[14180] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:09] DEBUG[14180] chan_sip.c: SIP call-id changed from '24e7688d7df856a427c7e0d045dd3abc@127.0.1.1:5060' to '71c549ac1adedf1f733725e63c013547@159.65.48.104:5060' [Aug 18 10:34:09] DEBUG[14180] stasis.c: Creating topic. name: channel:213099, detail: [Aug 18 10:34:09] DEBUG[14180] stasis.c: Topic 'channel:213099': 0x7f0c84139df0 created [Aug 18 10:34:09] DEBUG[14180] stasis.c: Creating topic. name: cache:331/channel:213099, detail: [Aug 18 10:34:09] DEBUG[14180] stasis.c: Topic 'cache:331/channel:213099': 0x7f0c840881b0 created [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:09] DEBUG[14087] channel.c: Channel 0x7f0c10131cc0 'SIP/zvonobot-00000080' allocated [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:09] DEBUG[14087] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:09] DEBUG[14087] res_stasis.c: calls_0: Subscribing to 213092 [Aug 18 10:34:09] DEBUG[14087] stasis/app.c: Channel '213092' is 1 interested in calls_0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[14111] channel.c: Channel 0x7f0c3c08c510 'Recorder/ARI-0000002c;1' allocated [Aug 18 10:34:09] DEBUG[14087] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:09] DEBUG[14087] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 (Checking To) --From tag as02885f54 --To-tag as5580464d [Aug 18 10:34:09] DEBUG[14111] stasis.c: Creating topic. name: channel:1629282849.285, detail: [Aug 18 10:34:09] DEBUG[14111] stasis.c: Topic 'channel:1629282849.285': 0x7f0c3c05f270 created [Aug 18 10:34:09] DEBUG[14111] stasis.c: Creating topic. name: cache:332/channel:1629282849.285, detail: [Aug 18 10:34:09] DEBUG[14111] stasis.c: Topic 'cache:332/channel:1629282849.285': 0x7f0c3c08bed0 created [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Outgoing Call for 79821116948 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '10507dcf059680b46ad884550335c862@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116988@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757 Max-Forwards: 70 From: ;tag=as02885f54 To: Contact: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #105 (3) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #105)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6316ms with no response [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[14205] chan_sip.c: Audio is at 11608 [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] VERBOSE[14205] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:09] DEBUG[13644] channel.c: Channel 0x7f0c2c090fb0 'SIP/zvonobot-00000052' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000052 - start 1629282838.953041 answer 0.000000 end 1629282849.488134 dur 10.535 bill 1629282849.488 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] VERBOSE[14205] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:09] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 From: ;tag=as0261f463 To: ;tag=as64b58d1a Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 460639390 460639390 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a920936;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0261f463 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64b58d1a [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 460639390 460639390 IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17848 RTP/AVP 0 8 101 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking To) --From tag as0261f463 --To-tag as64b58d1a [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:09] VERBOSE[14205] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Strict routing enforced for session 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:09] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:09] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117049@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK60e3ca6c Max-Forwards: 70 From: ;tag=as0261f463 To: ;tag=as64b58d1a Contact: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (6) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116974@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda Max-Forwards: 70 From: ;tag=as5b87d923 To: Contact: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 231516279 231516279 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:09] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (1) INVITE - 5 [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Initializing initreq for method INVITE - callid 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116948@178.62.121.41 SIP/2.0 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 3 [ 52]: From: ;tag=as0cd290ec [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 6 [ 60]: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:09 GMT [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] VERBOSE[14205] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #65 [Aug 18 10:34:09] DEBUG[14205] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13767] chan_sip.c: Hangup call SIP/zvonobot-0000005a, SIP callid 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13767] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[13767] res_rtp_asterisk.c: (0x7f0c2c08f640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13767] res_rtp_asterisk.c: (0x7f0c2c08f640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13767] channel.c: Channel 0x7f0c2c0a7290 'SIP/zvonobot-0000005a' destroying [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213056': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213056' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[13280] chan_sip.c: Hangup call SIP/zvonobot-00000036, SIP callid 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13280] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13280] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13280] channel.c: Channel 0x7f0c34047590 'SIP/zvonobot-00000036' destroying [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213020': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213020' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282849.286, detail: [Aug 18 10:34:09] DEBUG[13772] chan_sip.c: Hangup call SIP/zvonobot-0000005b, SIP callid 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13772] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[13772] res_rtp_asterisk.c: (0x7f0c28107140) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13772] res_rtp_asterisk.c: (0x7f0c28107140) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13772] channel.c: Channel 0x7f0c280d3fe0 'SIP/zvonobot-0000005b' destroying [Aug 18 10:34:09] DEBUG[14118] channel.c: Channel 0x7f0c7809b2b0 'Recorder/ARI-0000002d;1' allocated [Aug 18 10:34:09] DEBUG[14118] stasis.c: Creating topic. name: channel:1629282849.287, detail: [Aug 18 10:34:09] DEBUG[14118] stasis.c: Topic 'channel:1629282849.287': 0x7f0c78065f10 created [Aug 18 10:34:09] DEBUG[14207] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.286': 0x7f0c300ba000 created [Aug 18 10:34:09] DEBUG[14207] http.c: HTTP Request URI is /ari/channels/213056 [Aug 18 10:34:09] DEBUG[14118] stasis.c: Creating topic. name: cache:333/channel:1629282849.287, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: cache:334/channel:1629282849.286, detail: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[14205] dial.c: Called zvonobot/79821116948 [Aug 18 10:34:09] DEBUG[14207] http.c: match request [ari/channels/213056] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14207] http.c: match request [ari/channels/213056] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[13982] channel.c: Channel 0x7f0c280a4fb0 'Announcer/ARI-00000026;1' destroying [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:334/channel:1629282849.286': 0x7f0c30131780 created [Aug 18 10:34:09] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13992] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50' [Aug 18 10:34:09] DEBUG[13992] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[13992] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:09] DEBUG[14207] http.c: match request [ari/channels/213056] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[14207] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[14118] stasis.c: Topic 'cache:333/channel:1629282849.287': 0x7f0c7805f590 created [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213054': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213054' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: cache:334/channel:1629282849.286, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:334/channel:1629282849.286': 0x7f0c30131780 destroyed [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282849.286, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.286': 0x7f0c300ba000 destroyed [Aug 18 10:34:09] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:00', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005a', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213056', '')] [Aug 18 10:34:09] DEBUG[13944] bridge_channel.c: Setting 0x7f0c280d1290(Announcer/ARI-00000026;2) state from:0 to:1 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13280] stasis.c: Destroying topic. name: cache:90/channel:213020, detail: [Aug 18 10:34:09] DEBUG[13280] stasis.c: Topic 'cache:90/channel:213020': 0x7f0c34048ce0 destroyed [Aug 18 10:34:09] DEBUG[13280] stasis.c: Destroying topic. name: channel:213020, detail: [Aug 18 10:34:09] DEBUG[13280] stasis.c: Topic 'channel:213020': 0x7f0c340488e0 destroyed [Aug 18 10:34:09] DEBUG[14208] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[14208] http.c: HTTP Request URI is /ari/channels/213020 [Aug 18 10:34:09] DEBUG[14209] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Finding handler for channels/213056 [Aug 18 10:34:09] DEBUG[13944] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pulling 0x7f0c280d1290(Announcer/ARI-00000026;2) [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[13944] bridge_channel.c: Channel Announcer/ARI-00000026;2 left 'softmix' stasis-bridge [Aug 18 10:34:09] DEBUG[13944] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c280d1290(Announcer/ARI-00000026;2) is leaving softmix technology [Aug 18 10:34:09] DEBUG[13077] bridge_channel.c: Setting 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) state from:0 to:1 [Aug 18 10:34:09] DEBUG[13077] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: pulling 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) [Aug 18 10:34:09] VERBOSE[13077] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50394-0x7f0c18009d50 left 'simple_bridge' stasis-bridge <87d87304-31e6-4326-b367-680423189269> [Aug 18 10:34:09] DEBUG[13077] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) is leaving simple_bridge technology [Aug 18 10:34:09] DEBUG[13077] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269' can not use native RTP bridge as two channels are required [Aug 18 10:34:09] DEBUG[13077] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:09] DEBUG[13077] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13077] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13077] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:09] DEBUG[13077] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269 is already using the new technology. [Aug 18 10:34:09] DEBUG[13077] stasis/control.c: robot_212964, 87d87304-31e6-4326-b367-680423189269: Channel was departed from bridge [Aug 18 10:34:09] DEBUG[13077] stasis/app.c: bridge '87d87304-31e6-4326-b367-680423189269': is 3 interested in calls_0 [Aug 18 10:34:09] DEBUG[13077] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:09] DEBUG[14078] channel.c: Channel 0x7f0cb008f170 'SIP/zvonobot-00000081' allocated [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:09] DEBUG[14078] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:09] DEBUG[13944] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d'. Checking compatability for channels 'SIP/zvonobot-0000002a' and 'Recorder/ARI-00000020;2' [Aug 18 10:34:09] DEBUG[13072] stasis/control.c: robot_212964: Channel departing bridge [Aug 18 10:34:09] DEBUG[13072] bridge.c: Waiting for 0x7f0c2c028920(UnicastRTP/127.0.0.1:50394-0x7f0c18009d50) bridge thread to die. [Aug 18 10:34:09] DEBUG[13072] stasis/app.c: channel 'robot_212964': is 1 interested in calls_0 [Aug 18 10:34:09] DEBUG[13072] channel.c: Channel 0x7f0c18079740 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[13767] stasis.c: Destroying topic. name: cache:231/channel:213056, detail: [Aug 18 10:34:09] DEBUG[13767] stasis.c: Topic 'cache:231/channel:213056': 0x7f0c2c012da0 destroyed [Aug 18 10:34:09] DEBUG[13767] stasis.c: Destroying topic. name: channel:213056, detail: [Aug 18 10:34:09] DEBUG[13767] stasis.c: Topic 'channel:213056': 0x7f0c2c07bc70 destroyed [Aug 18 10:34:09] DEBUG[13944] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as channel 'SIP/zvonobot-0000002a' has features which prevent it [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13944] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:09] VERBOSE[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: switching from softmix technology to simple_bridge [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology constructor [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: moving 0x7f0c7006de00(SIP/zvonobot-0000002a) to dummy bridge temporarily [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: moving 0x7f0c2c08b700(Recorder/ARI-00000020;2) to dummy bridge temporarily [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is leaving softmix technology (dummy) [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is leaving softmix technology (dummy) [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling softmix technology stop [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is joining simple_bridge technology [Aug 18 10:34:09] DEBUG[13944] channel.c: Channel Recorder/ARI-00000020;2 setting read format path: slin -> slin [Aug 18 10:34:09] DEBUG[14211] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Finding handler for channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Finding handler for 213056 [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking channels create: Didn't match 213056 [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14207] res_ari.c: Checking channels externalMedia: Didn't match 213056 [Aug 18 10:34:09] DEBUG[14211] http.c: HTTP Request URI is /ari/channels/213054 [Aug 18 10:34:09] DEBUG[14211] http.c: match request [ari/channels/213054] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14207] res_ari.c: No explicit handler found for 213056. Using wildcard channelId. [Aug 18 10:34:09] DEBUG[14208] http.c: match request [ari/channels/213020] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14211] http.c: match request [ari/channels/213054] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[13944] channel.c: Channel Recorder/ARI-00000020;2 setting write format path: slin -> slin [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is joining simple_bridge technology [Aug 18 10:34:09] DEBUG[13944] channel.c: Channel Recorder/ARI-00000020;2 setting read format path: slin -> slin [Aug 18 10:34:09] DEBUG[13944] channel.c: Channel Recorder/ARI-00000020;2 setting write format path: slin -> slin [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling simple_bridge technology start [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: deferring softmix technology destructor [Aug 18 10:34:09] DEBUG[13944] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: queueing action type:13 sub:1000 [Aug 18 10:34:09] DEBUG[14209] http.c: HTTP Request URI is /ari/channels/212964 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:09] DEBUG[14208] http.c: match request [ari/channels/213020] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282849.288, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.288': 0x7f0c30131780 created [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: cache:335/channel:1629282849.288, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:335/channel:1629282849.288': 0x7f0c300ba000 created [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: cache:335/channel:1629282849.288, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:335/channel:1629282849.288': 0x7f0c300ba000 destroyed [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282849.288, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.288': 0x7f0c30131780 destroyed [Aug 18 10:34:09] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000036', '', 'AppDial2', '(Outgoing Line)', 14, 0, 'BUSY', 3, '', '213020', '')] [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31c5c295 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282849.289, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.289': 0x7f0c300ba000 created [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: cache:336/channel:1629282849.289, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:336/channel:1629282849.289': 0x7f0c300bc4c0 created [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag as31c5c295 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6445ms with no response [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14211] http.c: match request [ari/channels/213054] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13885] channel.c: Channel 0x7f0cac032dc0 'SIP/zvonobot-00000065' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[13772] stasis.c: Destroying topic. name: cache:232/channel:213054, detail: [Aug 18 10:34:09] DEBUG[13772] stasis.c: Topic 'cache:232/channel:213054': 0x7f0c280d6770 destroyed [Aug 18 10:34:09] DEBUG[13772] stasis.c: Destroying topic. name: channel:213054, detail: [Aug 18 10:34:09] DEBUG[13772] stasis.c: Topic 'channel:213054': 0x7f0c280d5d40 destroyed [Aug 18 10:34:09] DEBUG[14211] http.c: Match made with [ari] [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6358ms with no response [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[14208] http.c: match request [ari/channels/213020] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: cache:336/channel:1629282849.289, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:336/channel:1629282849.289': 0x7f0c300bc4c0 destroyed [Aug 18 10:34:09] DEBUG[14209] http.c: match request [ari/channels/212964] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282849.289, detail: [Aug 18 10:34:09] DEBUG[13982] stasis.c: Destroying topic. name: cache:254/channel:1629282842.214, detail: [Aug 18 10:34:09] DEBUG[13982] stasis.c: Topic 'cache:254/channel:1629282842.214': 0x7f0c280da0f0 destroyed [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.289': 0x7f0c300ba000 destroyed [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14208] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[13982] stasis.c: Destroying topic. name: channel:1629282842.214, detail: [Aug 18 10:34:09] DEBUG[13982] stasis.c: Topic 'channel:1629282842.214': 0x7f0c280d1ac0 destroyed [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Finding handler for channels/213054 [Aug 18 10:34:09] DEBUG[14209] http.c: match request [ari/channels/212964] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Finding handler for channels/213020 [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Finding handler for channels [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:09] DEBUG[13958] bridge_softmix.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: stopping mixing thread [Aug 18 10:34:09] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:00', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005b', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213054', '')] [Aug 18 10:34:09] DEBUG[20534] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:09] DEBUG[13666] channel.c: SIP/zvonobot-0000002a: Dropping redundant connected line update "" <>. [Aug 18 10:34:09] DEBUG[20534] bridge_softmix.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: Waiting for mixing thread to die. [Aug 18 10:34:09] DEBUG[14209] http.c: match request [ari/channels/212964] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 640, ms is 100 [Aug 18 10:34:09] DEBUG[13702] channel.c: Recorder/ARI-00000020;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6322ms with no response [Aug 18 10:34:09] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 736, ms is 112 [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13887] channel.c: Channel 0x7f0c10040760 'SIP/zvonobot-00000064' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[13944] channel.c: Channel 0x7f0c280acfd0 'Announcer/ARI-00000026;2' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Finding handler for channels [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212964-00000000 - start 1629282827.254584 answer 1629282827.254584 end 1629282849.598325 dur 22.343 bill 22.343 dispo ANSWERED [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000065 - start 1629282843.042776 answer 0.000000 end 1629282849.625224 dur 6.582 bill 1629282849.625 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000064 - start 1629282842.944457 answer 0.000000 end 1629282849.651542 dur 6.707 bill 1629282849.651 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14209] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[13648] channel.c: Channel 0x7f0cb00de970 'SIP/zvonobot-00000054' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:09] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000054 - start 1629282839.084666 answer 0.000000 end 1629282849.664942 dur 10.580 bill 1629282849.664 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Finding handler for 213054 [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c08f640) DTLS stop [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking channels create: Didn't match 213054 [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14211] res_ari.c: Checking channels externalMedia: Didn't match 213054 [Aug 18 10:34:09] DEBUG[14211] res_ari.c: No explicit handler found for 213054. Using wildcard channelId. [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Finding handler for channels/212964 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Finding handler for channels [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Finding handler for 213020 [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:09] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c08f640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c08f640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c08f640) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c08f640' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3403efe0) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3403efe0' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2596122845f5f4322466678f68967bbf@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c28107140) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c28107140) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c28107140) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c28107140) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking channels create: Didn't match 213020 [Aug 18 10:34:09] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:09] DEBUG[14208] res_ari.c: Checking channels externalMedia: Didn't match 213020 [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c28107140' [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6395ms with no response [Aug 18 10:34:09] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP got report of 76 bytes from 178.62.121.41:18113 [Aug 18 10:34:09] DEBUG[14208] res_ari.c: No explicit handler found for 213020. Using wildcard channelId. [Aug 18 10:34:09] DEBUG[14136] channel.c: Channel 0x7f0c9c09b130 'Announcer/ARI-0000002f;1' allocated [Aug 18 10:34:09] DEBUG[13996] stasis.c: Creating topic. name: channel:1629282849.290, detail: [Aug 18 10:34:09] DEBUG[13996] stasis.c: Topic 'channel:1629282849.290': 0x7f0c2000d150 created [Aug 18 10:34:09] DEBUG[13996] stasis.c: Creating topic. name: cache:337/channel:1629282849.290, detail: [Aug 18 10:34:09] DEBUG[13996] stasis.c: Topic 'cache:337/channel:1629282849.290': 0x7f0c20067350 created [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[14120] channel.c: Channel 0x7f0c80025890 'Recorder/ARI-0000002e;1' allocated [Aug 18 10:34:09] DEBUG[14120] stasis.c: Creating topic. name: channel:1629282849.292, detail: [Aug 18 10:34:09] DEBUG[14120] stasis.c: Topic 'channel:1629282849.292': 0x7f0c800417d0 created [Aug 18 10:34:09] DEBUG[14120] stasis.c: Creating topic. name: cache:338/channel:1629282849.292, detail: [Aug 18 10:34:09] DEBUG[14120] stasis.c: Topic 'cache:338/channel:1629282849.292': 0x7f0c800419b0 created [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:09] DEBUG[14136] stasis.c: Creating topic. name: channel:1629282849.291, detail: [Aug 18 10:34:09] DEBUG[14136] stasis.c: Topic 'channel:1629282849.291': 0x7f0c9c035ab0 created [Aug 18 10:34:09] DEBUG[14136] stasis.c: Creating topic. name: cache:339/channel:1629282849.291, detail: [Aug 18 10:34:09] DEBUG[14136] stasis.c: Topic 'cache:339/channel:1629282849.291': 0x7f0c9c0082e0 created [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Finding handler for 212964 [Aug 18 10:34:09] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:09] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking channels create: Didn't match 212964 [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14078] res_stasis.c: calls_0: Subscribing to 213091 [Aug 18 10:34:09] DEBUG[14078] stasis/app.c: Channel '213091' is 1 interested in calls_0 [Aug 18 10:34:09] DEBUG[13518] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13890] channel.c: Channel 0x7f0c1c12fa80 'SIP/zvonobot-00000066' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[14078] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:09] DEBUG[14078] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14209] res_ari.c: Checking channels externalMedia: Didn't match 212964 [Aug 18 10:34:09] DEBUG[14209] res_ari.c: No explicit handler found for 212964. Using wildcard channelId. [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000066 - start 1629282843.182427 answer 0.000000 end 1629282849.753785 dur 6.571 bill 1629282849.753 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Outgoing Call for 79821116949 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 From: ;tag=as366f0ed0 To: ;tag=as0a7e373f Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a7e373f [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 (Checking To) --From tag as366f0ed0 --To-tag as0a7e373f [Aug 18 10:34:09] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13470] app.c: One waitfor failed, trying another [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (1) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 10507dcf059680b46ad884550335c862@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6450ms with no response [Aug 18 10:34:09] WARNING[20585] chan_sip.c: Hanging up call 10507dcf059680b46ad884550335c862@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13655] channel.c: Channel 0x7f0c20081870 'SIP/zvonobot-00000056' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (4) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (3) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000056 - start 1629282839.217099 answer 0.000000 end 1629282849.820898 dur 10.603 bill 1629282849.820 dispo NO ANSWER [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #111 (5) INVITE - 5 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #111)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116958@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412 Max-Forwards: 70 From: ;tag=as15514e30 To: Contact: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1229396789 1229396789 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11576 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (4) INVITE - 5 [Aug 18 10:34:09] VERBOSE[14212] chan_sip.c: Audio is at 18068 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 From: ;tag=as009e460a To: ;tag=as64de9d5c Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 568221000 568221000 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13198 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as009e460a [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64de9d5c [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:09] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 568221000 568221000 IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13198 RTP/AVP 0 8 101 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:09] VERBOSE[14212] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:09] VERBOSE[14212] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:09] VERBOSE[14212] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Initializing initreq for method INVITE - callid 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116949@178.62.121.41 SIP/2.0 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 3 [ 52]: From: ;tag=as7d784780 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 6 [ 60]: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:09 GMT [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 (Checking To) --From tag as009e460a --To-tag as64de9d5c [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:09] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:09] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117018@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fbc007d Max-Forwards: 70 From: ;tag=as009e460a To: ;tag=as64de9d5c Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (4) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:09] VERBOSE[14212] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #116 [Aug 18 10:34:09] DEBUG[14212] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14044] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:09] DEBUG[14044] http.c: HTTP closing session. Top level [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[13801] chan_sip.c: Hangup call SIP/zvonobot-0000005f, SIP callid 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13801] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[13801] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13801] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13801] channel.c: Channel 0x7f0c78050c90 'SIP/zvonobot-0000005f' destroying [Aug 18 10:34:09] DEBUG[14139] channel.c: Channel 0x7f0c9808fde0 'UnicastRTP/127.0.0.1:50497-0x7f0c98083570' allocated [Aug 18 10:34:09] DEBUG[14139] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:09] VERBOSE[14139] res_rtp_asterisk.c: 0x7f0c98014a50 -- Strict RTP learning after remote address set to: 127.0.0.1:50497 [Aug 18 10:34:09] DEBUG[14139] res_stasis.c: calls_0: Subscribing to robot_212982 [Aug 18 10:34:09] DEBUG[14139] stasis/app.c: Channel 'robot_212982' is 1 interested in calls_0 [Aug 18 10:34:09] DEBUG[14155] channel.c: Channel 0x7f0c180999a0 'Recorder/ARI-00000030;1' allocated [Aug 18 10:34:09] DEBUG[14155] stasis.c: Creating topic. name: channel:1629282849.293, detail: [Aug 18 10:34:09] DEBUG[14155] stasis.c: Topic 'channel:1629282849.293': 0x7f0c180d88e0 created [Aug 18 10:34:09] DEBUG[14155] stasis.c: Creating topic. name: cache:340/channel:1629282849.293, detail: [Aug 18 10:34:09] DEBUG[14155] stasis.c: Topic 'cache:340/channel:1629282849.293': 0x7f0c180d8b10 created [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[13195] chan_sip.c: Hangup call SIP/zvonobot-0000002e, SIP callid 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13782] chan_sip.c: Hangup call SIP/zvonobot-0000005c, SIP callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13797] chan_sip.c: Hangup call SIP/zvonobot-0000005e, SIP callid 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13195] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13793] chan_sip.c: Hangup call SIP/zvonobot-0000005d, SIP callid 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[13195] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13468] channel.c: Channel 0x7f0c88086180 'Recorder/ARI-00000014;2' destroying [Aug 18 10:34:09] DEBUG[13797] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[13866] channel.c: Channel 0x7f0c8410b600 'Announcer/ARI-00000025;2' destroying [Aug 18 10:34:09] VERBOSE[13470] app.c: User hung up [Aug 18 10:34:09] DEBUG[13797] res_rtp_asterisk.c: (0x7f0c340ed1d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13671] channel.c: Channel 0x7f0c9c024160 'Announcer/ARI-0000001d;1' destroying [Aug 18 10:34:09] DEBUG[13797] res_rtp_asterisk.c: (0x7f0c340ed1d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13797] channel.c: Channel 0x7f0c340fb900 'SIP/zvonobot-0000005e' destroying [Aug 18 10:34:09] DEBUG[13793] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213058': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213058' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213055': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213055' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[13801] stasis.c: Destroying topic. name: cache:240/channel:213058, detail: [Aug 18 10:34:09] DEBUG[13801] stasis.c: Topic 'cache:240/channel:213058': 0x7f0c7803abb0 destroyed [Aug 18 10:34:09] DEBUG[13801] stasis.c: Destroying topic. name: channel:213058, detail: [Aug 18 10:34:09] DEBUG[13801] stasis.c: Topic 'channel:213058': 0x7f0c7803c810 destroyed [Aug 18 10:34:09] DEBUG[13518] bridge_channel.c: Setting 0x2c12c90(Snoop/213012-00000009) state from:0 to:1 [Aug 18 10:34:09] DEBUG[13518] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: pulling 0x2c12c90(Snoop/213012-00000009) [Aug 18 10:34:09] VERBOSE[13518] bridge_channel.c: Channel Snoop/213012-00000009 left 'simple_bridge' stasis-bridge [Aug 18 10:34:09] DEBUG[13518] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x2c12c90(Snoop/213012-00000009) is leaving simple_bridge technology [Aug 18 10:34:09] DEBUG[13518] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' can not use native RTP bridge as two channels are required [Aug 18 10:34:09] DEBUG[13518] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:09] DEBUG[13518] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13518] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:09] DEBUG[13518] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:09] DEBUG[13518] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 is already using the new technology. [Aug 18 10:34:09] DEBUG[13518] bridge_channel.c: Bridge is returning 0x2c12c90(Snoop/213012-00000009) to read format slin [Aug 18 10:34:09] DEBUG[13518] channel.c: Channel Snoop/213012-00000009 setting read format path: slin -> slin [Aug 18 10:34:09] DEBUG[13518] bridge_channel.c: Bridge is returning 0x2c12c90(Snoop/213012-00000009) to write format slin [Aug 18 10:34:09] DEBUG[13518] channel.c: Channel Snoop/213012-00000009 setting write format path: slin -> slin [Aug 18 10:34:09] DEBUG[13518] stasis/control.c: 1629282835.133, b7adaa29-9b73-48a7-8d8d-8ee58b870f71: Channel was departed from bridge [Aug 18 10:34:09] DEBUG[13518] stasis/app.c: bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71': is 3 interested in calls_0 [Aug 18 10:34:09] DEBUG[13518] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:09] DEBUG[13475] stasis/control.c: 1629282835.133: Channel departing bridge [Aug 18 10:34:09] DEBUG[13475] bridge.c: Waiting for 0x2c12c90(Snoop/213012-00000009) bridge thread to die. [Aug 18 10:34:09] DEBUG[13475] stasis/app.c: channel '1629282835.133': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[13475] stasis/app.c: channel '1629282835.133' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[13475] channel.c: Channel 0x7f0cac05bc90 'Snoop/213012-00000009' hanging up. Refs: 3 [Aug 18 10:34:09] DEBUG[13470] res_stasis_recording.c: 1629282835.130: Recording complete [Aug 18 10:34:09] DEBUG[13470] channel.c: Channel 0x7f0c88078fc0 'Recorder/ARI-00000014;1' hanging up. Refs: 2 [Aug 18 10:34:09] DEBUG[13793] res_rtp_asterisk.c: (0x7f0c38087180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13793] res_rtp_asterisk.c: (0x7f0c38087180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13793] channel.c: Channel 0x7f0c38056500 'SIP/zvonobot-0000005d' destroying [Aug 18 10:34:09] DEBUG[13671] stasis.c: Destroying topic. name: cache:204/channel:1629282838.171, detail: [Aug 18 10:34:09] DEBUG[13671] stasis.c: Topic 'cache:204/channel:1629282838.171': 0x7f0c9c032730 destroyed [Aug 18 10:34:09] DEBUG[13671] stasis.c: Destroying topic. name: channel:1629282838.171, detail: [Aug 18 10:34:09] DEBUG[13671] stasis.c: Topic 'channel:1629282838.171': 0x7f0c9c024d20 destroyed [Aug 18 10:34:09] DEBUG[13671] channel.c: Channel 0x7f0c9c05ac20 'Announcer/ARI-0000001d;2' destroying [Aug 18 10:34:09] DEBUG[13782] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:09] DEBUG[13782] res_rtp_asterisk.c: (0x7f0c40073870) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[13782] res_rtp_asterisk.c: (0x7f0c40073870) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[14139] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:09] DEBUG[14139] http.c: HTTP closing session. Top level [Aug 18 10:34:09] DEBUG[14219] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[13782] channel.c: Channel 0x7f0c4006f090 'SIP/zvonobot-0000005c' destroying [Aug 18 10:34:09] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282849.294, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.294': 0x7f0c300bc4c0 created [Aug 18 10:34:09] DEBUG[20545] stasis.c: Creating topic. name: cache:341/channel:1629282849.294, detail: [Aug 18 10:34:09] DEBUG[14216] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[14216] http.c: HTTP Request URI is /ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:09] DEBUG[14218] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:341/channel:1629282849.294': 0x7f0c300ba000 created [Aug 18 10:34:09] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[13866] stasis.c: Destroying topic. name: cache:253/channel:1629282842.213, detail: [Aug 18 10:34:09] DEBUG[13866] stasis.c: Topic 'cache:253/channel:1629282842.213': 0x7f0c840682a0 destroyed [Aug 18 10:34:09] DEBUG[13866] stasis.c: Destroying topic. name: channel:1629282842.213, detail: [Aug 18 10:34:09] DEBUG[13866] stasis.c: Topic 'channel:1629282842.213': 0x7f0c8407dfc0 destroyed [Aug 18 10:34:09] DEBUG[14216] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14216] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: cache:341/channel:1629282849.294, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'cache:341/channel:1629282849.294': 0x7f0c300ba000 destroyed [Aug 18 10:34:09] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282849.294, detail: [Aug 18 10:34:09] DEBUG[20545] stasis.c: Topic 'channel:1629282849.294': 0x7f0c300bc4c0 destroyed [Aug 18 10:34:09] DEBUG[13797] stasis.c: Destroying topic. name: cache:238/channel:213055, detail: [Aug 18 10:34:09] DEBUG[13797] stasis.c: Topic 'cache:238/channel:213055': 0x7f0c340fe760 destroyed [Aug 18 10:34:09] DEBUG[13797] stasis.c: Destroying topic. name: channel:213055, detail: [Aug 18 10:34:09] DEBUG[13797] stasis.c: Topic 'channel:213055': 0x7f0c340fdce0 destroyed [Aug 18 10:34:09] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213057': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213057' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis.c: Destroying topic. name: cache:235/channel:213057, detail: [Aug 18 10:34:09] DEBUG[20620] stasis.c: Topic 'cache:235/channel:213057': 0x7f0c38058c80 destroyed [Aug 18 10:34:09] DEBUG[20620] stasis.c: Destroying topic. name: channel:213057, detail: [Aug 18 10:34:09] DEBUG[20620] stasis.c: Topic 'channel:213057': 0x7f0c38058250 destroyed [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (2) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14219] http.c: HTTP Request URI is /ari/channels/213055 [Aug 18 10:34:09] DEBUG[14218] http.c: HTTP Request URI is /ari/channels/213058 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660 Max-Forwards: 70 From: ;tag=as6657c8e8 To: ;tag=as510b84fe Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="589c1c16", response="87884a26a92648e14764df9a3df9354f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[14216] http.c: match request [ari/bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14216] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[20523] threadpool.c: Increasing threadpool stasis/pool's size by 1 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Finding handler for bridges/e594e1d1-53fe-4904-8517-472d8e3b8b52/play [Aug 18 10:34:09] DEBUG[14218] http.c: match request [ari/channels/213058] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14220] http.c: HTTP opening session. Top level [Aug 18 10:34:09] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6657c8e8 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as510b84fe [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="589c1c16", response="87884a26a92648e14764df9a3df9354f" [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 (Checking From) --From tag as6657c8e8 --To-tag as510b84fe [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:09] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Received bye, no owner, selfdestruct soon. [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4569b660;received=178.62.121.41 From: ;tag=as6657c8e8 To: ;tag=as510b84fe Call-ID: 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (3) INVITE - 5 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2cc23538293c1849651dca44558c8447@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7806cff0' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ed1d0) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ed1d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ed1d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ed1d0) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c340ed1d0' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1ee655842d2ed684574010b3091c860a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38087180) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38087180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38087180) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38087180) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c38087180' [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Destroying SIP dialog 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40073870) DTLS stop [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40073870) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40073870) DTLS srtp - stopped timeout timer' [Aug 18 10:34:09] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40073870) ICE RTP transport deallocating [Aug 18 10:34:09] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c40073870' [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 From: ;tag=as3f810040 To: ;tag=as622102a0 Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" Content-Length: 0 <-------------> [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f948336;received=159.65.48.104 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f810040 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as622102a0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="416821cf" [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 (Checking To) --From tag as3f810040 --To-tag as622102a0 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Stopping retransmission on '50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:09] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116991@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64 Max-Forwards: 70 From: ;tag=as3f810040 To: Contact: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:09] DEBUG[14218] http.c: match request [ari/channels/213058] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Finding handler for bridges [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:09] DEBUG[14220] http.c: HTTP Request URI is /ari/channels/213057 [Aug 18 10:34:09] DEBUG[14218] http.c: match request [ari/channels/213058] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[14218] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:09] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005f', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213058', '')] [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213059': is 0 interested in calls_0 [Aug 18 10:34:09] DEBUG[20620] stasis/app.c: channel '213059' unsubscribed from calls_0 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:09] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:09] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14219] http.c: match request [ari/channels/213055] with handler [httpstatus] len 10 [Aug 18 10:34:09] VERBOSE[14215] dial.c: Called 127.0.0.1:50497 [Aug 18 10:34:09] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14220] http.c: match request [ari/channels/213057] with handler [httpstatus] len 10 [Aug 18 10:34:09] DEBUG[14220] http.c: match request [ari/channels/213057] with handler [phoneprov] len 9 [Aug 18 10:34:09] DEBUG[14220] http.c: match request [ari/channels/213057] with handler [ari] len 3 [Aug 18 10:34:09] DEBUG[14220] http.c: Match made with [ari] [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Finding handler for channels/213057 [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Finding handler for channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Finding handler for 213057 [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking channels create: Didn't match 213057 [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:09] DEBUG[14220] res_ari.c: Checking channels externalMedia: Didn't match 213057 [Aug 18 10:34:09] DEBUG[14220] res_ari.c: No explicit handler found for 213057. Using wildcard channelId. [Aug 18 10:34:09] DEBUG[14216] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Finding handler for e594e1d1-53fe-4904-8517-472d8e3b8b52 [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Finding handler for channels/213058 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 From: ;tag=as000dc064 To: ;tag=as67a63ec0 Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as000dc064 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as67a63ec0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 (Checking To) --From tag as000dc064 --To-tag as67a63ec0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 From: ;tag=as2618d3a5 To: ;tag=as6e1d4afe Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77f18dbf;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2618d3a5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e1d4afe [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72646695" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 (Checking To) --From tag as2618d3a5 --To-tag as6e1d4afe [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116994@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7 Max-Forwards: 70 From: ;tag=as2618d3a5 To: Contact: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13782] stasis.c: Destroying topic. name: cache:234/channel:213059, detail: [Aug 18 10:34:10] DEBUG[13782] stasis.c: Topic 'cache:234/channel:213059': 0x7f0c40071840 destroyed [Aug 18 10:34:10] DEBUG[13782] stasis.c: Destroying topic. name: channel:213059, detail: [Aug 18 10:34:10] DEBUG[13782] stasis.c: Topic 'channel:213059': 0x7f0c40070e10 destroyed [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] DEBUG[14219] http.c: match request [ari/channels/213055] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14219] http.c: match request [ari/channels/213055] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14219] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Finding handler for channels/213055 [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Finding handler for 213055 [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking channels create: Didn't match 213055 [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14219] res_ari.c: Checking channels externalMedia: Didn't match 213055 [Aug 18 10:34:10] DEBUG[14219] res_ari.c: No explicit handler found for 213055. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #116 (1) INVITE - 5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #116)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 From: ;tag=as6d7500c6 To: ;tag=as14883ee4 Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK46a5ace9;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6d7500c6 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as14883ee4 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b522bbc" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 (Checking To) --From tag as6d7500c6 --To-tag as14883ee4 [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Finding handler for 213058 [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking channels create: Didn't match 213058 [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14218] res_ari.c: Checking channels externalMedia: Didn't match 213058 [Aug 18 10:34:10] DEBUG[14218] res_ari.c: No explicit handler found for 213058. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:10] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 1520, ms is 115 [Aug 18 10:34:10] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14216] res_ari.c: No explicit handler found for e594e1d1-53fe-4904-8517-472d8e3b8b52. Using wildcard bridgeId. [Aug 18 10:34:10] DEBUG[13671] stasis.c: Destroying topic. name: cache:205/channel:1629282838.172, detail: [Aug 18 10:34:10] DEBUG[13671] stasis.c: Topic 'cache:205/channel:1629282838.172': 0x7f0c9c044450 destroyed [Aug 18 10:34:10] DEBUG[13671] stasis.c: Destroying topic. name: channel:1629282838.172, detail: [Aug 18 10:34:10] DEBUG[13671] stasis.c: Topic 'channel:1629282838.172': 0x7f0c9c001f00 destroyed [Aug 18 10:34:10] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 672, ms is 62 [Aug 18 10:34:10] DEBUG[14222] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:09] VERBOSE[14212] dial.c: Called zvonobot/79821116949 [Aug 18 10:34:10] DEBUG[14222] http.c: HTTP Request URI is /ari/channels/213059 [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116995@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK18e1213e Max-Forwards: 70 From: ;tag=as6d7500c6 To: Contact: Call-ID: 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP audio difference is 760, ms is 115 [Aug 18 10:34:10] DEBUG[14226] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Finding handler for play [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:10] DEBUG[14216] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14222] http.c: match request [ari/channels/213059] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14222] http.c: match request [ari/channels/213059] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14222] http.c: match request [ari/channels/213059] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14222] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Finding handler for channels/213059 [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.295, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.295': 0x7f0c300fdc80 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:342/channel:1629282850.295, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:342/channel:1629282850.295': 0x7f0c300fe4f0 created [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 From: ;tag=as14540915 To: ;tag=as3725d74e Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call leg/transaction does not exist [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5dedae5d;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14540915 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3725d74e [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 (Checking To) --From tag as14540915 --To-tag as3725d74e [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 61ef035b188984f8748196c5467a3ff9@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 474 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 From: ;tag=as52d5bd88 To: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01f7ec69;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as52d5bd88 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13347] res_rtp_asterisk.c: (0x7f0c40006350) RTP audio difference is 696, ms is 107 [Aug 18 10:34:10] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[14215] dial.c: UnicastRTP/127.0.0.1:50497-0x7f0c98083570 answered [Aug 18 10:34:10] VERBOSE[14215] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50497-0x7f0c98083570 [Aug 18 10:34:10] DEBUG[14215] stasis/app.c: Channel 'robot_212982' is 2 interested in calls_0 [Aug 18 10:34:10] DEBUG[14226] http.c: HTTP Request URI is /ari/channels/213104?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116936&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14226] http.c: match request [ari/channels/213104] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14226] http.c: match request [ari/channels/213104] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14226] http.c: match request [ari/channels/213104] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14226] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[13468] stasis.c: Destroying topic. name: cache:156/channel:1629282835.131, detail: [Aug 18 10:34:10] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:10] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTP audio difference is 688, ms is 106 [Aug 18 10:34:10] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 1072, ms is 87 [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13468] stasis.c: Topic 'cache:156/channel:1629282835.131': 0x7f0c88087db0 destroyed [Aug 18 10:34:10] DEBUG[13468] stasis.c: Destroying topic. name: channel:1629282835.131, detail: [Aug 18 10:34:10] DEBUG[13468] stasis.c: Topic 'channel:1629282835.131': 0x7f0c880747d0 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:342/channel:1629282850.295, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:342/channel:1629282850.295': 0x7f0c300fe4f0 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.295, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.295': 0x7f0c300fdc80 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213055', '')] [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060 (Checking To) --From tag as52d5bd88 --To-tag [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14237] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14237] http.c: HTTP Request URI is /ari/bridges/a76fe935-dd52-4012-a523-638ab1ec4dfe/addChannel?channel=1629282845.251%2Crobot_212982 [Aug 18 10:34:10] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 720, ms is 65 [Aug 18 10:34:10] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 1296, ms is 101 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afb6be559ca864d571d35807d0bcac7@159.65.48.104:5060' Request 103: Found [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP response 100 to standard invite [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Finding handler for 213059 [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking channels create: Didn't match 213059 [Aug 18 10:34:10] DEBUG[14237] http.c: match request [ari/bridges/a76fe935-dd52-4012-a523-638ab1ec4dfe/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14222] res_ari.c: Checking channels externalMedia: Didn't match 213059 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (6) INVITE - 5 [Aug 18 10:34:10] DEBUG[14237] http.c: match request [ari/bridges/a76fe935-dd52-4012-a523-638ab1ec4dfe/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14237] http.c: match request [ari/bridges/a76fe935-dd52-4012-a523-638ab1ec4dfe/addChannel] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:10] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14237] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Finding handler for bridges/a76fe935-dd52-4012-a523-638ab1ec4dfe/addChannel [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Finding handler for bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Finding handler for a76fe935-dd52-4012-a523-638ab1ec4dfe [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14237] res_ari.c: No explicit handler found for a76fe935-dd52-4012-a523-638ab1ec4dfe. Using wildcard bridgeId. [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Finding handler for addChannel [Aug 18 10:34:10] DEBUG[14237] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:10] DEBUG[14237] stasis/control.c: 1629282845.251: Sending channel add_to_bridge command [Aug 18 10:34:10] DEBUG[14239] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14239] http.c: HTTP Request URI is /ari/channels/213105?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116935&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14222] res_ari.c: No explicit handler found for 213059. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14236] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 736, ms is 66 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116971@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734 Max-Forwards: 70 From: ;tag=as611ff9f7 To: Contact: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1375473090 1375473090 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13558 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 From: ;tag=as3d872a68 To: ;tag=as578d3717 Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK284a0077;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as578d3717 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="068a7741" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag as578d3717 [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14226] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14132] bridge_roles.c: Roles did not exist on channel Snoop/212982-00000010 [Aug 18 10:34:10] DEBUG[14132] stasis/control.c: 1629282845.251: Adding to bridge a76fe935-dd52-4012-a523-638ab1ec4dfe [Aug 18 10:34:10] DEBUG[14132] stasis/app.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' is 1 interested in calls_0 [Aug 18 10:34:10] DEBUG[14244] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c90040640(Snoop/212982-00000010) is joining [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116990@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6 Max-Forwards: 70 From: ;tag=as3d872a68 To: Contact: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[14239] http.c: match request [ari/channels/213105] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14239] http.c: match request [ari/channels/213105] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14239] http.c: match request [ari/channels/213105] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14239] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:10] DEBUG[14239] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Finding handler for channels/213105 [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Finding handler for 213105 [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking channels create: Didn't match 213105 [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14239] res_ari.c: Checking channels externalMedia: Didn't match 213105 [Aug 18 10:34:10] DEBUG[14239] res_ari.c: No explicit handler found for 213105. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14236] http.c: HTTP Request URI is /ari/channels/213107?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116933&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14251] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 704, ms is 64 [Aug 18 10:34:10] DEBUG[14246] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[13695] audiohook.c: Audiohook 0x7f0cb011ab60 has stale audio in its factories. Flushing them both [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14251] http.c: HTTP Request URI is /ari/channels/213109?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116931&callerId=74950493843 [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (3) INVITE - 5 [Aug 18 10:34:10] DEBUG[14244] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: pushing 0x7f0c90040640(Snoop/212982-00000010) [Aug 18 10:34:10] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 864, ms is 74 [Aug 18 10:34:10] VERBOSE[14244] bridge_channel.c: Channel Snoop/212982-00000010 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:10] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 704, ms is 64 [Aug 18 10:34:10] DEBUG[14251] http.c: match request [ari/channels/213109] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14246] http.c: HTTP Request URI is /ari/channels/213106?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116934&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14246] http.c: match request [ari/channels/213106] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14246] http.c: match request [ari/channels/213106] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14246] http.c: match request [ari/channels/213106] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Finding handler for channels/213104 [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Finding handler for 213104 [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking channels create: Didn't match 213104 [Aug 18 10:34:10] DEBUG[14246] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14246] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Finding handler for channels/213106 [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Finding handler for 213106 [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking channels create: Didn't match 213106 [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14246] res_ari.c: Checking channels externalMedia: Didn't match 213106 [Aug 18 10:34:10] DEBUG[14246] res_ari.c: No explicit handler found for 213106. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14256] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14251] http.c: match request [ari/channels/213109] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14259] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14259] http.c: HTTP Request URI is /ari/channels/213111?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116929&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14226] res_ari.c: Checking channels externalMedia: Didn't match 213104 [Aug 18 10:34:10] DEBUG[14226] res_ari.c: No explicit handler found for 213104. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for Snoop/213012-00000009 - start 1629282835.580057 answer 1629282835.580057 end 1629282849.936503 dur 14.356 bill 14.356 dispo ANSWERED [Aug 18 10:34:10] DEBUG[14244] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' can not use native RTP bridge as two channels are required [Aug 18 10:34:10] DEBUG[14244] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:10] DEBUG[14244] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14244] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14244] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:10] DEBUG[14244] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe is already using the new technology. [Aug 18 10:34:10] DEBUG[14256] http.c: HTTP Request URI is /ari/channels/213108?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116932&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14236] http.c: match request [ari/channels/213107] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14259] http.c: match request [ari/channels/213111] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14256] http.c: match request [ari/channels/213108] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14256] http.c: match request [ari/channels/213108] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14256] http.c: match request [ari/channels/213108] with handler [ari] len 3 [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6445ms with no response [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14256] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14256] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Finding handler for channels/213108 [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Finding handler for 213108 [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking channels create: Didn't match 213108 [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14256] res_ari.c: Checking channels externalMedia: Didn't match 213108 [Aug 18 10:34:10] DEBUG[14256] res_ari.c: No explicit handler found for 213108. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14244] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c90040640(Snoop/212982-00000010) is joining simple_bridge technology [Aug 18 10:34:10] DEBUG[14251] http.c: match request [ari/channels/213109] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14236] http.c: match request [ari/channels/213107] with handler [phoneprov] len 9 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1498731843 1498731843 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18334 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as31e40966 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20bcc5bd [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[14170] channel.c: Channel 0x7f0c7c013ef0 'SIP/zvonobot-00000082' allocated [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:10] DEBUG[14170] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:10] DEBUG[14170] res_stasis.c: calls_0: Subscribing to 213096 [Aug 18 10:34:10] DEBUG[14170] stasis/app.c: Channel '213096' is 1 interested in calls_0 [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14236] http.c: match request [ari/channels/213107] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[13920] channel.c: Channel 0x7f0c24110ce0 'SIP/zvonobot-00000069' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.296, detail: [Aug 18 10:34:10] DEBUG[14251] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14170] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.296': 0x7f0c3007f570 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:343/channel:1629282850.296, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:343/channel:1629282850.296': 0x7f0c300fd3c0 created [Aug 18 10:34:10] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1498731843 1498731843 IN IP4 178.62.121.41 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18334 RTP/AVP 0 8 101 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking To) --From tag as31e40966 --To-tag as20bcc5bd [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Got SDP version 1498731843 and unique parts [root 1498731843 IN IP4 178.62.121.41] [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1498731843 1498731843 IN IP4 178.62.121.41... OK. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) ICE set role failed; no ice instance [Aug 18 10:34:10] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) RTCP setting address on RTP instance [Aug 18 10:34:10] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c84032110 -- Strict RTP learning after remote address set to: 178.62.121.41:18334 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:18334 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00f0d48) from 0x7f0c147e2330 to 0x7f0c8402e6f8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00d3da8) from 0x7f0c147e2330 to 0x7f0c8402e6f8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0036cc8) from 0x7f0c147e2330 to 0x7f0c8402e6f8 [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) RTCP ignoring duplicate property [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:10] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001f setting read format path: alaw -> alaw [Aug 18 10:34:10] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000001f setting write format path: alaw -> alaw [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS - ast_rtp_activate rtp=0x7f0c84032110 - setup and perform DTLS' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84032110) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84032110) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:10] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Strict routing enforced for session 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117044@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c9cb4c4 Max-Forwards: 70 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #26 (4) INVITE - 5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #26)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #116 (2) INVITE - 5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #116)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Session timer started: 40 - 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 1768000ms [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7;received=159.65.48.104 From: ;tag=as3da39e97 To: ;tag=as4a709074 Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3da7eb51" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3da39e97 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4a709074 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3da7eb51" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 (Checking To) --From tag as3da39e97 --To-tag as4a709074 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #105 (4) INVITE - 5 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:343/channel:1629282850.296, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:343/channel:1629282850.296': 0x7f0c300fd3c0 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.296, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.296': 0x7f0c3007f570 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005d', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213057', '')] [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Outgoing Call for 79821116944 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14170] http.c: HTTP closing session. Top level [Aug 18 10:34:10] VERBOSE[13109] dial.c: SIP/zvonobot-0000001f answered [Aug 18 10:34:10] VERBOSE[13109] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000001f [Aug 18 10:34:10] DEBUG[14236] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:10] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13109] stasis/app.c: Channel '212996' is 2 interested in calls_0 [Aug 18 10:34:10] DEBUG[14215] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14266] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14266] http.c: HTTP Request URI is /ari/channels/213113?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116927&callerId=74950493843 [Aug 18 10:34:10] DEBUG[14266] http.c: match request [ari/channels/213113] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14266] http.c: match request [ari/channels/213113] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14266] http.c: match request [ari/channels/213113] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14266] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14266] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Finding handler for channels/213113 [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Finding handler for 213113 [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking channels create: Didn't match 213113 [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14266] res_ari.c: Checking channels externalMedia: Didn't match 213113 [Aug 18 10:34:10] DEBUG[14260] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14266] res_ari.c: No explicit handler found for 213113. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #105)) [Aug 18 10:34:10] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14259] http.c: match request [ari/channels/213111] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14132] stasis/app.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' is 2 interested in calls_0 [Aug 18 10:34:10] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 10 instead [Aug 18 10:34:10] DEBUG[14263] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14259] http.c: match request [ari/channels/213111] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14259] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14259] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14263] http.c: HTTP Request URI is /ari/channels/213110?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116930&callerId=74950493843 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.297, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.297': 0x7f0c3007f570 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:344/channel:1629282850.297, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:344/channel:1629282850.297': 0x7f0c30122290 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:344/channel:1629282850.297, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:344/channel:1629282850.297': 0x7f0c30122290 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.297, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.297': 0x7f0c3007f570 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000005c', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213059', '')] [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Finding handler for channels/213111 [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14237] stasis/control.c: robot_212982: Sending channel add_to_bridge command [Aug 18 10:34:10] DEBUG[14263] http.c: match request [ari/channels/213110] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Finding handler for 213111 [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking channels create: Didn't match 213111 [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14259] res_ari.c: Checking channels externalMedia: Didn't match 213111 [Aug 18 10:34:10] DEBUG[14259] res_ari.c: No explicit handler found for 213111. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000069 - start 1629282843.447674 answer 0.000000 end 1629282850.217414 dur 6.769 bill 1629282850.217 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[14263] http.c: match request [ari/channels/213110] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: Allocating new SIP dialog for 33ddd8c471b8714157cf1c150e1c5e35@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14251] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Finding handler for channels/213109 [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14246] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c0924e0' [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) RTP allocated port 19144 [Aug 18 10:34:10] DEBUG[14236] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14263] http.c: match request [ari/channels/213110] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Finding handler for 213109 [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking channels create: Didn't match 213109 [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14251] res_ari.c: Checking channels externalMedia: Didn't match 213109 [Aug 18 10:34:10] DEBUG[14251] res_ari.c: No explicit handler found for 213109. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE creating session 0.0.0.0:19144 (19144) [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE create [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14246] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14246] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE add candidate: 159.65.48.104:19144, 2130706431 [Aug 18 10:34:10] DEBUG[14246] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14246] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE add candidate: 10.131.0.10:19144, 2130706431 [Aug 18 10:34:10] DEBUG[14246] rtp_engine.c: RTP instance '0x7f0c7c0924e0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) ICE stopped [Aug 18 10:34:10] DEBUG[14246] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14246] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14246] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 From: ;tag=as404b2233 To: ;tag=as0706ba37 Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as404b2233 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0706ba37 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 (Checking To) --From tag as404b2233 --To-tag as0706ba37 [Aug 18 10:34:10] VERBOSE[14262] chan_sip.c: Audio is at 14552 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:10] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Finding handler for channels/213107 [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Finding handler for 213107 [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking channels create: Didn't match 213107 [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14236] res_ari.c: Checking channels externalMedia: Didn't match 213107 [Aug 18 10:34:10] DEBUG[14236] res_ari.c: No explicit handler found for 213107. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14246] res_rtp_asterisk.c: (0x7f0c7c0924e0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14246] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14246] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14246] chan_sip.c: SIP call-id changed from '33ddd8c471b8714157cf1c150e1c5e35@127.0.1.1:5060' to '137837c51322c444587a45b5059337ee@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14246] stasis.c: Creating topic. name: channel:213106, detail: [Aug 18 10:34:10] DEBUG[14246] stasis.c: Topic 'channel:213106': 0x7f0c7c03dc40 created [Aug 18 10:34:10] DEBUG[14246] stasis.c: Creating topic. name: cache:345/channel:213106, detail: [Aug 18 10:34:10] DEBUG[14246] stasis.c: Topic 'cache:345/channel:213106': 0x7f0c7c0c4e90 created [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117052@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e Max-Forwards: 70 From: ;tag=as404b2233 To: ;tag=as0706ba37 Contact: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6542ms with no response [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[13041] dial.c: SIP/zvonobot-0000001a is busy [Aug 18 10:34:10] DEBUG[13041] channel.c: Channel 0x7f0c78021200 'SIP/zvonobot-0000001a' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6543ms with no response [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[14262] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (3) INVITE - 5 [Aug 18 10:34:10] DEBUG[13658] channel.c: Channel 0x7f0c28104e60 'SIP/zvonobot-00000058' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14260] http.c: HTTP Request URI is /ari/channels/213112?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116928&callerId=74950493843 [Aug 18 10:34:10] DEBUG[13939] channel.c: Channel 0x7f0c180bb1b0 'SIP/zvonobot-00000067' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: Allocating new SIP dialog for 2b91b4410886e096037f31223f079a9a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14263] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6495ms with no response [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001a - start 1629282826.166696 answer 0.000000 end 1629282850.408832 dur 24.242 bill 1629282850.408 dispo BUSY [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: Allocating new SIP dialog for 3cdd7f6c0c22d9da5131d024209e79b3@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[14262] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:10] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP got report of 100 bytes from 178.62.121.41:10913 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[13661] channel.c: Channel 0x7f0cac079f80 'SIP/zvonobot-00000055' hanging up. Refs: 3 [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6386ms with no response [Aug 18 10:34:10] DEBUG[13704] audiohook.c: Audiohook 0x7f0c8805b620 has stale audio in its factories. Flushing them both [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[14260] http.c: match request [ari/channels/213112] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14263] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: Allocating new SIP dialog for 420ccdda4644be1e45b11171214651b1@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14226] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c082ca0' [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) RTP allocated port 15928 [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE creating session 0.0.0.0:15928 (15928) [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE create [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14226] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14226] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE add candidate: 159.65.48.104:15928, 2130706431 [Aug 18 10:34:10] DEBUG[14226] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14226] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE add candidate: 10.131.0.10:15928, 2130706431 [Aug 18 10:34:10] DEBUG[14226] rtp_engine.c: RTP instance '0x7f0c3c082ca0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) ICE stopped [Aug 18 10:34:10] DEBUG[14226] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14226] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14226] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14226] res_rtp_asterisk.c: (0x7f0c3c082ca0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14226] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14226] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[13950] channel.c: Channel 0x7f0c080ee420 'SIP/zvonobot-0000006b' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[14260] http.c: match request [ari/channels/213112] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14215] stasis/control.c: robot_212982: Adding to bridge a76fe935-dd52-4012-a523-638ab1ec4dfe [Aug 18 10:34:10] DEBUG[14215] stasis/app.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' is 3 interested in calls_0 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Finding handler for channels/213110 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[14262] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:10] DEBUG[14272] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) is joining [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14256] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c84126bb0' [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14260] http.c: match request [ari/channels/213112] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 752, ms is 67 [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000058 - start 1629282839.318784 answer 0.000000 end 1629282850.419340 dur 11.100 bill 1629282850.419 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000067 - start 1629282843.838657 answer 0.000000 end 1629282850.424954 dur 6.586 bill 1629282850.424 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000055 - start 1629282839.295706 answer 0.000000 end 1629282850.443115 dur 11.147 bill 1629282850.443 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006b - start 1629282843.962126 answer 0.000000 end 1629282850.460808 dur 6.498 bill 1629282850.460 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) RTP allocated port 14160 [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14260] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 From: ;tag=as39a2ec19 To: ;tag=as70b1d74e Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1138223289 1138223289 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11310 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39a2ec19 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70b1d74e [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[14260] http.c: HTTP consuming request body [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Finding handler for channels/213112 [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Finding handler for 213110 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14239] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c74055f50' [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking channels create: Didn't match 213110 [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14263] res_ari.c: Checking channels externalMedia: Didn't match 213110 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Initializing initreq for method INVITE - callid 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116944@178.62.121.41 SIP/2.0 [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Finding handler for 213112 [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE creating session 0.0.0.0:14160 (14160) [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE create [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14256] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14256] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE add candidate: 159.65.48.104:14160, 2130706431 [Aug 18 10:34:10] DEBUG[14256] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14256] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE add candidate: 10.131.0.10:14160, 2130706431 [Aug 18 10:34:10] DEBUG[14256] rtp_engine.c: RTP instance '0x7f0c84126bb0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE stopped [Aug 18 10:34:10] DEBUG[14256] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14256] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14256] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14256] res_rtp_asterisk.c: (0x7f0c84126bb0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14256] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14256] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1138223289 1138223289 IN IP4 178.62.121.41 [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking channels create: Didn't match 213112 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 3 [ 52]: From: ;tag=as22d5765f [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14260] res_ari.c: Checking channels externalMedia: Didn't match 213112 [Aug 18 10:34:10] DEBUG[14260] res_ari.c: No explicit handler found for 213112. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14263] res_ari.c: No explicit handler found for 213110. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11310 RTP/AVP 0 8 101 [Aug 18 10:34:10] DEBUG[14226] chan_sip.c: SIP call-id changed from '420ccdda4644be1e45b11171214651b1@127.0.1.1:5060' to '1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14226] stasis.c: Creating topic. name: channel:213104, detail: [Aug 18 10:34:10] DEBUG[14226] stasis.c: Topic 'channel:213104': 0x7f0c3c10a460 created [Aug 18 10:34:10] DEBUG[14226] stasis.c: Creating topic. name: cache:346/channel:213104, detail: [Aug 18 10:34:10] DEBUG[14226] stasis.c: Topic 'cache:346/channel:213104': 0x7f0c3c017d70 created [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14272] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: pushing 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 (Checking To) --From tag as39a2ec19 --To-tag as70b1d74e [Aug 18 10:34:10] DEBUG[14256] chan_sip.c: SIP call-id changed from '3cdd7f6c0c22d9da5131d024209e79b3@127.0.1.1:5060' to '747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14256] stasis.c: Creating topic. name: channel:213108, detail: [Aug 18 10:34:10] DEBUG[14256] stasis.c: Topic 'channel:213108': 0x7f0c84149770 created [Aug 18 10:34:10] DEBUG[14256] stasis.c: Creating topic. name: cache:347/channel:213108, detail: [Aug 18 10:34:10] DEBUG[14256] stasis.c: Topic 'cache:347/channel:213108': 0x7f0c8414a1b0 created [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) RTP allocated port 14706 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 6 [ 60]: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE creating session 0.0.0.0:14706 (14706) [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:10 GMT [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: Allocating new SIP dialog for 75fdbb325787c08b620d63d03779a027@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: Allocating new SIP dialog for 05a1b426213182e172de8a3a517d966c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Stopping retransmission on '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Got SDP version 1138223289 and unique parts [root 1138223289 IN IP4 178.62.121.41] [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1138223289 1138223289 IN IP4 178.62.121.41... OK. [Aug 18 10:34:10] DEBUG[14115] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE create [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[14251] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7806cff0' [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: Allocating new SIP dialog for 4e3794e56bb5090918204f323a887af7@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14236] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38078a20' [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) ICE set role failed; no ice instance [Aug 18 10:34:10] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) RTCP setting address on RTP instance [Aug 18 10:34:10] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c3400da00 -- Strict RTP learning after remote address set to: 178.62.121.41:11310 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:11310 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb007d878) from 0x7f0c147e2330 to 0x7f0c34009fe8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0143ae8) from 0x7f0c147e2330 to 0x7f0c34009fe8 [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0076bb8) from 0x7f0c147e2330 to 0x7f0c34009fe8 [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE add system candidates [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) RTCP ignoring duplicate property [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:10] VERBOSE[14262] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #33 [Aug 18 10:34:10] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000006 setting read format path: alaw -> alaw [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) RTP allocated port 11680 [Aug 18 10:34:10] DEBUG[14262] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000006 setting write format path: alaw -> alaw [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) DTLS - ast_rtp_activate rtp=0x7f0c3400da00 - setup and perform DTLS' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3400da00) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3400da00) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:10] DEBUG[14182] channel.c: Channel 0x7f0c80068db0 'SIP/zvonobot-00000083' allocated [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:10] VERBOSE[14272] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:10] DEBUG[14182] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:10] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:10] DEBUG[14259] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c80061fd0' [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) RTP allocated port 10398 [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:10] DEBUG[13870] channel.c: Channel 0x7f0c90025910 'Announcer/ARI-0000002b;2' allocated [Aug 18 10:34:10] DEBUG[13870] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:10] DEBUG[13870] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000002b;1' [Aug 18 10:34:10] DEBUG[13810] chan_sip.c: Hangup call SIP/zvonobot-00000060, SIP callid 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13810] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:10] DEBUG[13810] res_rtp_asterisk.c: (0x7f0c90052d80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13810] res_rtp_asterisk.c: (0x7f0c90052d80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13810] channel.c: Channel 0x7f0c9007e3b0 'SIP/zvonobot-00000060' destroying [Aug 18 10:34:10] DEBUG[13636] chan_sip.c: Hangup call SIP/zvonobot-00000051, SIP callid 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13636] res_rtp_asterisk.c: (0x7f0c100f9ed0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13636] res_rtp_asterisk.c: (0x7f0c100f9ed0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13636] channel.c: Channel 0x7f0c10106720 'SIP/zvonobot-00000051' destroying [Aug 18 10:34:10] DEBUG[13631] chan_sip.c: Hangup call SIP/zvonobot-00000050, SIP callid 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13811] chan_sip.c: Hangup call SIP/zvonobot-00000062, SIP callid 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13631] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13811] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:10] DEBUG[13811] res_rtp_asterisk.c: (0x7f0c84090800) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13631] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13811] res_rtp_asterisk.c: (0x7f0c84090800) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13631] channel.c: Channel 0x7f0cb40628c0 'SIP/zvonobot-00000050' destroying [Aug 18 10:34:10] DEBUG[13811] channel.c: Channel 0x7f0c841031a0 'SIP/zvonobot-00000062' destroying [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:10] DEBUG[14182] res_stasis.c: calls_0: Subscribing to 213098 [Aug 18 10:34:10] DEBUG[14182] stasis/app.c: Channel '213098' is 1 interested in calls_0 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) RTP allocated port 19502 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.301, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.301': 0x7f0c3007f570 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:348/channel:1629282850.301, detail: [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Strict routing enforced for session 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:10] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:10] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117071@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b681b4c Max-Forwards: 70 From: ;tag=as39a2ec19 To: ;tag=as70b1d74e Contact: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[12894] dial.c: SIP/zvonobot-00000006 answered [Aug 18 10:34:10] DEBUG[14182] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14273] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c900b05e0(Announcer/ARI-0000002b;2) is joining [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14182] http.c: HTTP closing session. Top level [Aug 18 10:34:10] VERBOSE[12894] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000006 [Aug 18 10:34:10] DEBUG[12894] stasis/app.c: Channel '212969' is 2 interested in calls_0 [Aug 18 10:34:10] VERBOSE[12894] res_rtp_asterisk.c: 0x7f0c3400da00 -- Strict RTP switching to RTP target address 178.62.121.41:11310 as source [Aug 18 10:34:10] DEBUG[12894] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:10] VERBOSE[14262] dial.c: Called zvonobot/79821116944 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Outgoing Call for 79821116942 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:348/channel:1629282850.301': 0x7f0c3005a840 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:348/channel:1629282850.301, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:348/channel:1629282850.301': 0x7f0c3005a840 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.301, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.301': 0x7f0c3007f570 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:58', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000051', '', 'AppDial2', '(Outgoing Line)', 9, 0, 'NO ANSWER', 3, '', '213047', '')] [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.302, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.302': 0x7f0c3007f570 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:349/channel:1629282850.302, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:349/channel:1629282850.302': 0x7f0c300e5490 created [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213047': is 0 interested in calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213047' unsubscribed from calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: cache:190/channel:213047, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'cache:190/channel:213047': 0x7f0c10108ef0 destroyed [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: channel:213047, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'channel:213047': 0x7f0c10108470 destroyed [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE creating session 0.0.0.0:10398 (10398) [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE creating session 0.0.0.0:11680 (11680) [Aug 18 10:34:10] DEBUG[12894] channel.c: Channel SIP/zvonobot-00000006 setting read format path: ulaw -> alaw [Aug 18 10:34:10] DEBUG[12894] channel.c: Channel SIP/zvonobot-00000006 setting write format path: alaw -> ulaw [Aug 18 10:34:10] DEBUG[14115] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14115] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14115] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14115] channel.c: Channel Announcer/ARI-00000027;1 setting write format path: slin -> slin [Aug 18 10:34:10] DEBUG[14115] channel.c: Channel 0x7f0c980440c0 'Announcer/ARI-00000027;1' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[14272] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe'. Checking compatability for channels 'Snoop/212982-00000010' and 'UnicastRTP/127.0.0.1:50497-0x7f0c98083570' [Aug 18 10:34:10] DEBUG[14272] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' can not use native RTP bridge as could not get details [Aug 18 10:34:10] DEBUG[14272] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:10] DEBUG[14272] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14272] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14272] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:10] DEBUG[14272] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe is already using the new technology. [Aug 18 10:34:10] DEBUG[14272] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) is joining simple_bridge technology [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 setting read format path: slin16 -> slin16 [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel Snoop/212982-00000010 setting write format path: slin16 -> slin [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213062': is 0 interested in calls_0 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213062' unsubscribed from calls_0 [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2485aced650f4f671041baca16773141@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6506ms with no response [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: cache:241/channel:213062, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'cache:241/channel:213062': 0x7f0c90081290 destroyed [Aug 18 10:34:10] WARNING[20585] chan_sip.c: Hanging up call 2485aced650f4f671041baca16773141@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:349/channel:1629282850.302, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:349/channel:1629282850.302': 0x7f0c300e5490 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.302, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.302': 0x7f0c3007f570 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000060', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213062', '')] [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: channel:213062, detail: [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE creating session 0.0.0.0:19502 (19502) [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'channel:213062': 0x7f0c90080810 destroyed [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE create [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE create [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (5) INVITE - 5 [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50497-0x7f0c98083570 - start 1629282849.922534 answer 1629282850.055776 end 1629282850.634171 dur 0.711 bill 0.578 dispo ANSWERED [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116959@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a Max-Forwards: 70 From: ;tag=as3ecc0b7c To: Contact: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1559812355 1559812355 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11206 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:10] VERBOSE[14274] chan_sip.c: Audio is at 15826 [Aug 18 10:34:10] VERBOSE[14274] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (4) INVITE - 5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:10] DEBUG[14239] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14259] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14259] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE add candidate: 159.65.48.104:19502, 2130706431 [Aug 18 10:34:10] DEBUG[14259] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14259] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE add candidate: 10.131.0.10:19502, 2130706431 [Aug 18 10:34:10] DEBUG[14259] rtp_engine.c: RTP instance '0x7f0c80061fd0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE stopped [Aug 18 10:34:10] DEBUG[14259] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14259] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[13953] channel.c: Channel 0x7f0c3010c400 'SIP/zvonobot-0000006c' hanging up. Refs: 2 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (5) INVITE - 5 [Aug 18 10:34:10] VERBOSE[14274] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE create [Aug 18 10:34:10] DEBUG[14239] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213060': is 0 interested in calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213060' unsubscribed from calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: cache:243/channel:213060, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'cache:243/channel:213060': 0x7f0c84105b60 destroyed [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: channel:213060, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'channel:213060': 0x7f0c841050e0 destroyed [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel Snoop/212982-00000010 setting read format path: slin -> slin16 [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 setting write format path: slin16 -> slin16 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213048': is 0 interested in calls_0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213048' unsubscribed from calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: cache:189/channel:213048, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'cache:189/channel:213048': 0x7f0cb4065270 destroyed [Aug 18 10:34:10] DEBUG[14259] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14259] res_rtp_asterisk.c: (0x7f0c80061fd0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14259] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116963@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38d2b3ca Max-Forwards: 70 From: ;tag=as27ec27d0 To: Contact: Call-ID: 564726e17074235c1af6801638e43e42@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 268674259 268674259 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14548 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE add candidate: 159.65.48.104:14706, 2130706431 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: channel:213048, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'channel:213048': 0x7f0cb404b1e0 destroyed [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (5) INVITE - 5 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116957@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09ad65ea Max-Forwards: 70 From: ;tag=as4bf88154 To: Contact: Call-ID: 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1867117864 1867117864 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18420 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] VERBOSE[14274] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.303, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.303': 0x7f0c300e5490 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:350/channel:1629282850.303, detail: [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE add system candidates [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #116 (3) INVITE - 5 [Aug 18 10:34:10] DEBUG[14259] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14273] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pushing 0x7f0c900b05e0(Announcer/ARI-0000002b;2) [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:10] DEBUG[14251] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14236] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14236] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE add candidate: 159.65.48.104:10398, 2130706431 [Aug 18 10:34:10] DEBUG[14236] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14236] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE add candidate: 10.131.0.10:10398, 2130706431 [Aug 18 10:34:10] DEBUG[14236] rtp_engine.c: RTP instance '0x7f0c38078a20' is setup and ready to go [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) ICE stopped [Aug 18 10:34:10] DEBUG[14236] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14236] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14236] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14236] res_rtp_asterisk.c: (0x7f0c38078a20) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14236] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14236] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14236] chan_sip.c: SIP call-id changed from '4e3794e56bb5090918204f323a887af7@127.0.1.1:5060' to '5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14251] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add candidate: 159.65.48.104:11680, 2130706431 [Aug 18 10:34:10] DEBUG[14251] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14251] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE add candidate: 10.131.0.10:11680, 2130706431 [Aug 18 10:34:10] DEBUG[14251] rtp_engine.c: RTP instance '0x7f0c7806cff0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE stopped [Aug 18 10:34:10] DEBUG[14251] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14251] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14251] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14251] res_rtp_asterisk.c: (0x7f0c7806cff0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14251] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #116)) [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Session timer started: 118 - 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 1768000ms [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14236] stasis.c: Creating topic. name: channel:213107, detail: [Aug 18 10:34:10] DEBUG[14236] stasis.c: Topic 'channel:213107': 0x7f0c3807ef40 created [Aug 18 10:34:10] DEBUG[14236] stasis.c: Creating topic. name: cache:351/channel:213107, detail: [Aug 18 10:34:10] DEBUG[14236] stasis.c: Topic 'cache:351/channel:213107': 0x7f0c380939c0 created [Aug 18 10:34:10] DEBUG[14251] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Initializing initreq for method INVITE - callid 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14239] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14239] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE add candidate: 10.131.0.10:14706, 2130706431 [Aug 18 10:34:10] DEBUG[14239] rtp_engine.c: RTP instance '0x7f0c74055f50' is setup and ready to go [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) ICE stopped [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116942@178.62.121.41 SIP/2.0 [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:350/channel:1629282850.303': 0x7f0c30142830 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:350/channel:1629282850.303, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:350/channel:1629282850.303': 0x7f0c30142830 destroyed [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 3 [ 52]: From: ;tag=as001c84c2 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 6 [ 60]: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:10 GMT [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:10] VERBOSE[14274] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #69 [Aug 18 10:34:10] DEBUG[14274] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:10] VERBOSE[14274] dial.c: Called zvonobot/79821116942 [Aug 18 10:34:10] DEBUG[14215] stasis/app.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' is 4 interested in calls_0 [Aug 18 10:34:10] DEBUG[14237] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:10] DEBUG[14237] http.c: HTTP closing session. Top level [Aug 18 10:34:10] DEBUG[14275] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14278] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14278] http.c: HTTP Request URI is /ari/channels/213047 [Aug 18 10:34:10] DEBUG[14275] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.303, detail: [Aug 18 10:34:10] DEBUG[14279] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 13 instead [Aug 18 10:34:10] DEBUG[14159] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14279] http.c: HTTP Request URI is /ari/channels/213062 [Aug 18 10:34:10] DEBUG[14251] chan_sip.c: SIP call-id changed from '75fdbb325787c08b620d63d03779a027@127.0.1.1:5060' to '577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14259] chan_sip.c: SIP call-id changed from '05a1b426213182e172de8a3a517d966c@127.0.1.1:5060' to '217865353a22dc3331285fc05bb15812@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.303': 0x7f0c300e5490 destroyed [Aug 18 10:34:10] DEBUG[14279] http.c: match request [ari/channels/213062] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14273] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:10] VERBOSE[14273] bridge_channel.c: Channel Announcer/ARI-0000002b;2 joined 'simple_bridge' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:34:10] DEBUG[13819] chan_sip.c: Hangup call SIP/zvonobot-00000061, SIP callid 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[13819] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:10] DEBUG[13819] res_rtp_asterisk.c: (0x7f0c88063530) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[13819] res_rtp_asterisk.c: (0x7f0c88063530) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[14279] http.c: match request [ari/channels/213062] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14279] http.c: match request [ari/channels/213062] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14279] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14259] stasis.c: Creating topic. name: channel:213111, detail: [Aug 18 10:34:10] DEBUG[14278] http.c: match request [ari/channels/213047] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14282] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[13819] channel.c: Channel 0x7f0c8809bb30 'SIP/zvonobot-00000061' destroying [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Finding handler for channels/213062 [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Finding handler for 213062 [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking channels create: Didn't match 213062 [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14279] res_ari.c: Checking channels externalMedia: Didn't match 213062 [Aug 18 10:34:10] DEBUG[14279] res_ari.c: No explicit handler found for 213062. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14259] stasis.c: Topic 'channel:213111': 0x7f0c80030290 created [Aug 18 10:34:10] DEBUG[14259] stasis.c: Creating topic. name: cache:352/channel:213111, detail: [Aug 18 10:34:10] DEBUG[14259] stasis.c: Topic 'cache:352/channel:213111': 0x7f0c800317c0 created [Aug 18 10:34:10] DEBUG[14251] stasis.c: Creating topic. name: channel:213109, detail: [Aug 18 10:34:10] DEBUG[14251] stasis.c: Topic 'channel:213109': 0x7f0c7807b2f0 created [Aug 18 10:34:10] DEBUG[14251] stasis.c: Creating topic. name: cache:353/channel:213109, detail: [Aug 18 10:34:10] DEBUG[14251] stasis.c: Topic 'cache:353/channel:213109': 0x7f0c7802e7f0 created [Aug 18 10:34:10] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14275] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 From: ;tag=as76ba9cfd To: ;tag=as041dcfd6 Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[14283] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000062', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213060', '')] [Aug 18 10:34:10] DEBUG[14275] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[14239] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14278] http.c: match request [ari/channels/213047] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14281] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14283] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as76ba9cfd [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as041dcfd6 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[14281] http.c: HTTP Request URI is /ari/channels/213060 [Aug 18 10:34:10] DEBUG[14275] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14278] http.c: match request [ari/channels/213047] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: Allocating new SIP dialog for 23440d6a4bcb48ae02d7e2ef2297d3e8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14260] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c104f10' [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) RTP allocated port 15988 [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE creating session 0.0.0.0:15988 (15988) [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE create [Aug 18 10:34:10] DEBUG[14282] http.c: HTTP Request URI is /ari/channels/213048 [Aug 18 10:34:10] DEBUG[14275] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14283] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE add system candidates [Aug 18 10:34:10] DEBUG[14260] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14260] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14278] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[14239] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" [Aug 18 10:34:10] DEBUG[14281] http.c: match request [ari/channels/213060] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14282] http.c: match request [ari/channels/213048] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[14239] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 (Checking To) --From tag as76ba9cfd --To-tag as041dcfd6 [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE add candidate: 159.65.48.104:15988, 2130706431 [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Finding handler for channels/213047 [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Finding handler for bridges [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14282] http.c: match request [ari/channels/213048] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14176] channel.c: Channel 0x7f0c780382e0 'SIP/zvonobot-00000084' allocated [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:10] DEBUG[14176] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:10] DEBUG[14176] res_stasis.c: calls_0: Subscribing to 213097 [Aug 18 10:34:10] DEBUG[14176] stasis/app.c: Channel '213097' is 1 interested in calls_0 [Aug 18 10:34:10] DEBUG[14281] http.c: match request [ari/channels/213060] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213063': is 0 interested in calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis/app.c: channel '213063' unsubscribed from calls_0 [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: cache:242/channel:213063, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'cache:242/channel:213063': 0x7f0c8809e6d0 destroyed [Aug 18 10:34:10] DEBUG[20620] stasis.c: Destroying topic. name: channel:213063, detail: [Aug 18 10:34:10] DEBUG[20620] stasis.c: Topic 'channel:213063': 0x7f0c8809dca0 destroyed [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Finding handler for bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:10] DEBUG[14275] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:10] DEBUG[14275] stasis.c: Creating topic. name: bridge:94ef4fc9-246b-4999-9567-b41f8ba44681, detail: [Aug 18 10:34:10] DEBUG[14275] stasis.c: Topic 'bridge:94ef4fc9-246b-4999-9567-b41f8ba44681': 0x7f0ca0073d00 created [Aug 18 10:34:10] DEBUG[14275] stasis.c: Creating topic. name: cache:354/bridge:94ef4fc9-246b-4999-9567-b41f8ba44681, detail: [Aug 18 10:34:10] DEBUG[14275] stasis.c: Topic 'cache:354/bridge:94ef4fc9-246b-4999-9567-b41f8ba44681': 0x7f0ca00df4a0 created [Aug 18 10:34:10] DEBUG[14275] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:10] DEBUG[14275] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:10] DEBUG[14275] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:10] DEBUG[14275] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:10] DEBUG[14275] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:10] DEBUG[14275] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: calling simple_bridge technology constructor [Aug 18 10:34:10] DEBUG[14275] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: calling simple_bridge technology start [Aug 18 10:34:10] DEBUG[14273] bridge.c: Chose bridge technology softmix [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[14176] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:10] DEBUG[14282] http.c: match request [ari/channels/213048] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14282] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[14283] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:10] VERBOSE[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: switching from simple_bridge technology to softmix [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: Allocating new SIP dialog for 29c4f9552884dd053f7a47f0705b1619@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14176] http.c: HTTP closing session. Top level [Aug 18 10:34:10] DEBUG[14260] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14260] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE add candidate: 10.131.0.10:15988, 2130706431 [Aug 18 10:34:10] DEBUG[14260] rtp_engine.c: RTP instance '0x7f0c8c104f10' is setup and ready to go [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE stopped [Aug 18 10:34:10] DEBUG[14260] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14260] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14260] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14260] res_rtp_asterisk.c: (0x7f0c8c104f10) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14260] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14260] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14260] chan_sip.c: SIP call-id changed from '23440d6a4bcb48ae02d7e2ef2297d3e8@127.0.1.1:5060' to '07f82ab44968293544eb273a476d91c1@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14260] stasis.c: Creating topic. name: channel:213112, detail: [Aug 18 10:34:10] DEBUG[14260] stasis.c: Topic 'channel:213112': 0x7f0c8c11e8c0 created [Aug 18 10:34:10] DEBUG[14260] stasis.c: Creating topic. name: cache:355/channel:213112, detail: [Aug 18 10:34:10] DEBUG[14260] stasis.c: Topic 'cache:355/channel:213112': 0x7f0c8c11ffc0 created [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (4) INVITE - 5 [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14275] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: Allocating new SIP dialog for 73eafa362fe48c2172eda6111a7c0b4a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:10] DEBUG[14266] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c90058ee0' [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) RTP allocated port 12990 [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE creating session 0.0.0.0:12990 (12990) [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE create [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE add system candidates [Aug 18 10:34:10] DEBUG[14266] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14266] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE add candidate: 159.65.48.104:12990, 2130706431 [Aug 18 10:34:10] DEBUG[14266] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14266] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE add candidate: 10.131.0.10:12990, 2130706431 [Aug 18 10:34:10] DEBUG[14266] rtp_engine.c: RTP instance '0x7f0c90058ee0' is setup and ready to go [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE stopped [Aug 18 10:34:10] DEBUG[14266] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14266] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14266] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14266] res_rtp_asterisk.c: (0x7f0c90058ee0) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14266] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14266] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14266] chan_sip.c: SIP call-id changed from '73eafa362fe48c2172eda6111a7c0b4a@127.0.1.1:5060' to '70757ec224866cc54887d48e040f5301@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14266] stasis.c: Creating topic. name: channel:213113, detail: [Aug 18 10:34:10] DEBUG[14266] stasis.c: Topic 'channel:213113': 0x7f0c900666d0 created [Aug 18 10:34:10] DEBUG[14266] stasis.c: Creating topic. name: cache:356/channel:213113, detail: [Aug 18 10:34:10] DEBUG[14266] stasis.c: Topic 'cache:356/channel:213113': 0x7f0c9007e730 created [Aug 18 10:34:10] DEBUG[14275] http.c: HTTP closing session. Top level [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology constructor [Aug 18 10:34:10] DEBUG[14263] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c940c4d70' [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Outgoing Call for 79821116943 [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c7803b960(SIP/zvonobot-0000002b) to dummy bridge temporarily [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.307, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.307': 0x7f0c3002cab0 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:357/channel:1629282850.307, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:357/channel:1629282850.307': 0x7f0c300e54b0 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:357/channel:1629282850.307, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:357/channel:1629282850.307': 0x7f0c300e54b0 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.307, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.307': 0x7f0c3002cab0 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:58', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000050', '', 'AppDial2', '(Outgoing Line)', 9, 0, 'NO ANSWER', 3, '', '213048', '')] [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) RTP allocated port 15638 [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE creating session 0.0.0.0:15638 (15638) [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE create [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE add system candidates [Aug 18 10:34:10] DEBUG[14263] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:10] DEBUG[14263] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE add candidate: 159.65.48.104:15638, 2130706431 [Aug 18 10:34:10] DEBUG[14263] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:10] DEBUG[14263] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:10] DEBUG[14286] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14239] res_rtp_asterisk.c: (0x7f0c74055f50) RTCP setup on RTP instance [Aug 18 10:34:10] DEBUG[14272] channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 setting write format path: slin -> slin16 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (1) INVITE - 5 [Aug 18 10:34:10] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) to dummy bridge temporarily [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE add candidate: 10.131.0.10:15638, 2130706431 [Aug 18 10:34:10] DEBUG[14263] rtp_engine.c: RTP instance '0x7f0c940c4d70' is setup and ready to go [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE stopped [Aug 18 10:34:10] DEBUG[14263] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:10] DEBUG[14263] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:10] DEBUG[14263] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:10] DEBUG[14263] res_rtp_asterisk.c: (0x7f0c940c4d70) RTCP setup on RTP instance [Aug 18 10:34:10] VERBOSE[14263] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14263] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14263] chan_sip.c: SIP call-id changed from '29c4f9552884dd053f7a47f0705b1619@127.0.1.1:5060' to '279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14263] stasis.c: Creating topic. name: channel:213110, detail: [Aug 18 10:34:10] DEBUG[14263] stasis.c: Topic 'channel:213110': 0x7f0c940d3430 created [Aug 18 10:34:10] DEBUG[14263] stasis.c: Creating topic. name: cache:358/channel:213110, detail: [Aug 18 10:34:10] DEBUG[14263] stasis.c: Topic 'cache:358/channel:213110': 0x7f0c940d3e70 created [Aug 18 10:34:10] DEBUG[14281] http.c: match request [ari/channels/213060] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP audio difference is 848, ms is 126 [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is leaving simple_bridge technology (dummy) [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:10] DEBUG[14287] http.c: HTTP opening session. Top level [Aug 18 10:34:10] DEBUG[14286] http.c: HTTP Request URI is /ari/channels/213063 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[14239] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:10] DEBUG[14239] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:10] DEBUG[14239] chan_sip.c: SIP call-id changed from '2b91b4410886e096037f31223f079a9a@127.0.1.1:5060' to '63e4041b488585c57e57de141ed1835f@159.65.48.104:5060' [Aug 18 10:34:10] DEBUG[14239] stasis.c: Creating topic. name: channel:213105, detail: [Aug 18 10:34:10] DEBUG[14239] stasis.c: Topic 'channel:213105': 0x7f0c7405c390 created [Aug 18 10:34:10] DEBUG[14239] stasis.c: Creating topic. name: cache:359/channel:213105, detail: [Aug 18 10:34:10] DEBUG[14239] stasis.c: Topic 'cache:359/channel:213105': 0x7f0c7405ce10 created [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology stop [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c900b05e0(Announcer/ARI-0000002b;2) is joining softmix technology [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Announcer/ARI-0000002b;2: [Aug 18 10:34:10] DEBUG[14273] channel.c: Channel Announcer/ARI-0000002b;2 setting write format path: slin -> slin [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Announcer/ARI-0000002b;2: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Announcer/ARI-0000002b;2: Not in SFU mode [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining softmix technology [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: SIP/zvonobot-0000002b: [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: SIP/zvonobot-0000002b: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: SIP/zvonobot-0000002b: Not in SFU mode [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining softmix technology [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Recorder/ARI-0000001a;2: [Aug 18 10:34:10] DEBUG[14273] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:10] DEBUG[14273] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Recorder/ARI-0000001a;2: [Aug 18 10:34:10] DEBUG[14273] bridge_softmix.c: Recorder/ARI-0000001a;2: Not in SFU mode [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology start [Aug 18 10:34:10] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology destructor [Aug 18 10:34:10] DEBUG[14272] res_rtp_asterisk.c: (0x7f0c98083570) RTP ooh, format changed from none to slin16 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:10] VERBOSE[14285] chan_sip.c: Audio is at 18438 [Aug 18 10:34:10] VERBOSE[14285] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:10] VERBOSE[14285] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:10] VERBOSE[14285] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Initializing initreq for method INVITE - callid 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116943@178.62.121.41 SIP/2.0 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 3 [ 52]: From: ;tag=as307f6396 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 6 [ 60]: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:10 GMT [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:10] VERBOSE[14285] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #122 [Aug 18 10:34:10] DEBUG[14285] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[14095] res_rtp_asterisk.c: 0x7f0cb400c820 -- Strict RTP learning complete - Locking on source address 178.62.121.41:14926 [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Finding handler for 213047 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking channels create: Didn't match 213047 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Finding handler for channels/213048 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (5) INVITE - 5 [Aug 18 10:34:10] DEBUG[14283] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:10] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:10] DEBUG[14281] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14159] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14159] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14159] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:10] DEBUG[14159] channel.c: Channel Announcer/ARI-00000028;1 setting write format path: slin -> slin [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116960@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776 Max-Forwards: 70 From: ;tag=as1180a433 To: Contact: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612217756 612217756 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14278] res_ari.c: Checking channels externalMedia: Didn't match 213047 [Aug 18 10:34:10] DEBUG[14278] res_ari.c: No explicit handler found for 213047. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14287] http.c: HTTP Request URI is /ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/addChannel?channel=212996 [Aug 18 10:34:10] DEBUG[14286] http.c: match request [ari/channels/213063] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[14159] channel.c: Channel 0x7f0c1c127e20 'Announcer/ARI-00000028;1' hanging up. Refs: 2 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14283] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:10] VERBOSE[14285] dial.c: Called zvonobot/79821116943 [Aug 18 10:34:10] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Destroying SIP dialog 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '083441fd621bd040753e952c5d9a1860@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90052d80) DTLS stop [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90052d80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90052d80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90052d80) ICE RTP transport deallocating [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c90052d80' [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Destroying SIP dialog 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f9ed0) DTLS stop [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f9ed0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f9ed0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f9ed0) ICE RTP transport deallocating [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c100f9ed0' [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Destroying SIP dialog 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '596ef69a675656833620d8345b47f33c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS stop [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) ICE RTP transport deallocating [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb4045900' [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Destroying SIP dialog 195b29ec6362148262de07066ce29e57@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[14287] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:10] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '195b29ec6362148262de07066ce29e57@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Finding handler for bridges [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Finding handler for bridges [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP got report of 76 bytes from 178.62.121.41:14927 [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84090800) DTLS stop [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Finding handler for channels/213060 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84090800) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[14286] http.c: match request [ari/channels/213063] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:10] DEBUG[14287] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:10] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006c - start 1629282843.997021 answer 0.000000 end 1629282850.718511 dur 6.721 bill 1629282850.718 dispo NO ANSWER [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282850.312, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.312': 0x7f0c3006dc40 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Creating topic. name: cache:360/channel:1629282850.312, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:360/channel:1629282850.312': 0x7f0c3007c9d0 created [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: cache:360/channel:1629282850.312, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'cache:360/channel:1629282850.312': 0x7f0c3007c9d0 destroyed [Aug 18 10:34:10] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282850.312, detail: [Aug 18 10:34:10] DEBUG[20545] stasis.c: Topic 'channel:1629282850.312': 0x7f0c3006dc40 destroyed [Aug 18 10:34:10] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:02', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000061', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213063', '')] [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Finding handler for channels [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84090800) DTLS srtp - stopped timeout timer' [Aug 18 10:34:10] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:10] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:10] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84090800) ICE RTP transport deallocating [Aug 18 10:34:10] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c84090800' [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Finding handler for 213048 [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:10] DEBUG[14286] http.c: match request [ari/channels/213063] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking channels create: Didn't match 213048 [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 From: ;tag=as000dc064 To: ;tag=as67a63ec0 Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" Content-Length: 0 <-------------> [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as000dc064 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as67a63ec0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:10] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: = Looking for Call ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 (Checking To) --From tag as000dc064 --To-tag as67a63ec0 [Aug 18 10:34:10] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14283] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:10] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Finding handler for 213060 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: Checking channels externalMedia: Didn't match 213048 [Aug 18 10:34:10] DEBUG[14283] stasis.c: Creating topic. name: bridge:fa1a4da9-c446-4fa8-95aa-bada67702e1d, detail: [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking channels create: Didn't match 213060 [Aug 18 10:34:10] DEBUG[14287] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/addChannel] with handler [ari] len 3 [Aug 18 10:34:10] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:10] DEBUG[14283] stasis.c: Topic 'bridge:fa1a4da9-c446-4fa8-95aa-bada67702e1d': 0x2c430b0 created [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:10] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:10] DEBUG[14282] res_ari.c: No explicit handler found for 213048. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14281] res_ari.c: Checking channels externalMedia: Didn't match 213060 [Aug 18 10:34:10] DEBUG[14286] http.c: Match made with [ari] [Aug 18 10:34:10] DEBUG[14281] res_ari.c: No explicit handler found for 213060. Using wildcard channelId. [Aug 18 10:34:10] DEBUG[14287] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (1) INVITE - 5 [Aug 18 10:34:10] DEBUG[14283] stasis.c: Creating topic. name: cache:361/bridge:fa1a4da9-c446-4fa8-95aa-bada67702e1d, detail: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:11] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Finding handler for channels/213063 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14288] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: starting mixing thread [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[13870] res_stasis_playback.c: 1629282846.257: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14283] stasis.c: Topic 'cache:361/bridge:fa1a4da9-c446-4fa8-95aa-bada67702e1d': 0x2c7dce0 created [Aug 18 10:34:11] DEBUG[14283] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as two channels are required [Aug 18 10:34:11] DEBUG[14283] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[13870] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:11] DEBUG[13870] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14283] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14283] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:11] DEBUG[14283] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14283] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: calling simple_bridge technology constructor [Aug 18 10:34:11] DEBUG[14283] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: calling simple_bridge technology start [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Finding handler for bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/addChannel [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:11] DEBUG[14283] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:11] DEBUG[14290] channel.c: Channel Announcer/ARI-0000002b;1 setting write format path: gsm -> slin [Aug 18 10:34:11] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP audio difference is 44232, ms is 5549 [Aug 18 10:34:11] DEBUG[14290] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:11] DEBUG[14283] http.c: HTTP closing session. Top level [Aug 18 10:34:11] VERBOSE[14290] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:11] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14291] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14291] http.c: HTTP Request URI is /ari/playbacks/b0b5abc4-ed07-4866-b13c-00b9c88afb30 [Aug 18 10:34:11] DEBUG[14291] http.c: match request [ari/playbacks/b0b5abc4-ed07-4866-b13c-00b9c88afb30] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14292] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:11] DEBUG[14291] http.c: match request [ari/playbacks/b0b5abc4-ed07-4866-b13c-00b9c88afb30] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (4) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:11] DEBUG[14292] http.c: HTTP Request URI is /ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/addChannel?channel=212969 [Aug 18 10:34:11] DEBUG[14291] http.c: match request [ari/playbacks/b0b5abc4-ed07-4866-b13c-00b9c88afb30] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14291] http.c: Match made with [ari] [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14292] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14292] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Finding handler for playbacks/b0b5abc4-ed07-4866-b13c-00b9c88afb30 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[13820] chan_sip.c: Hangup call SIP/zvonobot-00000063, SIP callid 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13820] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:11] DEBUG[13820] res_rtp_asterisk.c: (0x7f0c8c072bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13820] res_rtp_asterisk.c: (0x7f0c8c072bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13820] channel.c: Channel 0x7f0c8c0f9790 'SIP/zvonobot-00000063' destroying [Aug 18 10:34:11] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Finding handler for playbacks [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:11] DEBUG[14292] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/addChannel] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:11] DEBUG[14091] stasis.c: Creating topic. name: channel:1629282851.313, detail: [Aug 18 10:34:11] DEBUG[14091] stasis.c: Topic 'channel:1629282851.313': 0x7f0c180c3a70 created [Aug 18 10:34:11] DEBUG[14091] stasis.c: Creating topic. name: cache:362/channel:1629282851.313, detail: [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:11] DEBUG[14091] stasis.c: Topic 'cache:362/channel:1629282851.313': 0x7f0c180f0250 created [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213061': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213061' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.314, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.314': 0x7f0c3006dc40 created [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:363/channel:1629282851.314, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:363/channel:1629282851.314': 0x7f0c3002cab0 created [Aug 18 10:34:11] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Finding handler for b0b5abc4-ed07-4866-b13c-00b9c88afb30 [Aug 18 10:34:11] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:34:11] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:244/channel:213061, detail: [Aug 18 10:34:11] DEBUG[14291] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14291] res_ari.c: No explicit handler found for b0b5abc4-ed07-4866-b13c-00b9c88afb30. Using wildcard playbackId. [Aug 18 10:34:11] DEBUG[14291] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:11] DEBUG[14293] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:244/channel:213061': 0x7f0c8c00eca0 destroyed [Aug 18 10:34:11] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213061, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213061': 0x7f0c8c00eb50 destroyed [Aug 18 10:34:11] DEBUG[14291] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:363/channel:1629282851.314, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:363/channel:1629282851.314': 0x7f0c3002cab0 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.314, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.314': 0x7f0c3006dc40 destroyed [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:02', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000063', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213061', '')] [Aug 18 10:34:11] DEBUG[14290] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:11] DEBUG[14290] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:11] DEBUG[14290] channel.c: Channel Announcer/ARI-0000002b;1 setting write format path: slin -> slin [Aug 18 10:34:11] NOTICE[14290] res_stasis_playback.c: 1629282846.257: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:11] DEBUG[14292] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14294] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14293] http.c: HTTP Request URI is /ari/channels/213061 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:11] DEBUG[14290] channel.c: Channel 0x7f0c900818f0 'Announcer/ARI-0000002b;1' hanging up. Refs: 2 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Finding handler for bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/addChannel [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Finding handler for 213063 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:11] DEBUG[14294] http.c: HTTP Request URI is /ari/channels/robot_213008 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14293] http.c: match request [ari/channels/213061] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking channels create: Didn't match 213063 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14293] http.c: match request [ari/channels/213061] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:11] DEBUG[14293] http.c: match request [ari/channels/213061] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14286] res_ari.c: Checking channels externalMedia: Didn't match 213063 [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6425ms with no response [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:11] DEBUG[14293] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:11] DEBUG[14294] http.c: match request [ari/channels/robot_213008] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:11] DEBUG[14286] res_ari.c: No explicit handler found for 213063. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Finding handler for fa1a4da9-c446-4fa8-95aa-bada67702e1d [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Finding handler for channels/213061 [Aug 18 10:34:11] DEBUG[14294] http.c: match request [ari/channels/robot_213008] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Hanging up call 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Finding handler for 213061 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: No explicit handler found for fa1a4da9-c446-4fa8-95aa-bada67702e1d. Using wildcard bridgeId. [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking channels create: Didn't match 213061 [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14294] http.c: match request [ari/channels/robot_213008] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14293] res_ari.c: Checking channels externalMedia: Didn't match 213061 [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Finding handler for addChannel [Aug 18 10:34:11] DEBUG[14293] res_ari.c: No explicit handler found for 213061. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14294] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:11] DEBUG[14292] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Finding handler for channels/robot_213008 [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:11] DEBUG[14167] channel.c: Channel 0x7f0c7404c830 'SIP/zvonobot-00000085' allocated [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14167] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[12894] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000006 [Aug 18 10:34:11] DEBUG[14292] stasis/control.c: 212969: Sending channel add_to_bridge command [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Finding handler for 94ef4fc9-246b-4999-9567-b41f8ba44681 [Aug 18 10:34:11] DEBUG[12894] stasis/control.c: 212969: Adding to bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d [Aug 18 10:34:11] DEBUG[13981] channel.c: Channel 0x7f0c38099400 'SIP/zvonobot-0000006d' hanging up. Refs: 2 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006d - start 1629282844.260099 answer 0.000000 end 1629282851.156336 dur 6.896 bill 1629282851.156 dispo NO ANSWER [Aug 18 10:34:11] DEBUG[12894] stasis/app.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14287] res_ari.c: No explicit handler found for 94ef4fc9-246b-4999-9567-b41f8ba44681. Using wildcard bridgeId. [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88063530) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88063530) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88063530) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88063530) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c88063530' [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Finding handler for addChannel [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Finding handler for robot_213008 [Aug 18 10:34:11] DEBUG[14287] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking channels create: Didn't match robot_213008 [Aug 18 10:34:11] DEBUG[14183] channel.c: Channel 0x7f0c8c10a180 'SIP/zvonobot-00000086' allocated [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14294] res_ari.c: Checking channels externalMedia: Didn't match robot_213008 [Aug 18 10:34:11] DEBUG[14294] res_ari.c: No explicit handler found for robot_213008. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14167] res_stasis.c: calls_0: Subscribing to 213095 [Aug 18 10:34:11] DEBUG[14167] stasis/app.c: Channel '213095' is 1 interested in calls_0 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2;received=159.65.48.104 From: ;tag=as00c25c39 To: ;tag=as1eec528d Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4edcab50" Content-Length: 0 <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as00c25c39 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1eec528d [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4edcab50" [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 (Checking To) --From tag as00c25c39 --To-tag as1eec528d [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (4) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6428ms with no response [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Hanging up call 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14167] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14167] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[13659] channel.c: Channel 0x7f0c180d2b30 'SIP/zvonobot-00000057' hanging up. Refs: 2 [Aug 18 10:34:11] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000057 - start 1629282839.268529 answer 0.000000 end 1629282851.204867 dur 11.936 bill 1629282851.204 dispo NO ANSWER [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Outgoing Call for 79821116945 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6296ms with no response [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Hanging up call 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c072bb0) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c072bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c072bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13669] channel.c: Channel 0x7f0c340bc830 'SIP/zvonobot-00000059' hanging up. Refs: 2 [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c072bb0) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000059 - start 1629282839.512568 answer 0.000000 end 1629282851.207134 dur 11.694 bill 1629282851.207 dispo NO ANSWER [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8c072bb0' [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14183] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[14295] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c300a0130(SIP/zvonobot-00000006) is joining [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] VERBOSE[14296] chan_sip.c: Audio is at 14556 [Aug 18 10:34:11] DEBUG[14287] stasis/control.c: 212996: Sending channel add_to_bridge command [Aug 18 10:34:11] VERBOSE[14296] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] DEBUG[14183] res_stasis.c: calls_0: Subscribing to 213103 [Aug 18 10:34:11] VERBOSE[14296] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP got report of 76 bytes from 178.62.121.41:10695 [Aug 18 10:34:11] VERBOSE[14296] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] VERBOSE[14112] res_rtp_asterisk.c: 0x7f0c2403c460 -- Strict RTP learning complete - Locking on source address 178.62.121.41:10694 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14183] stasis/app.c: Channel '213103' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Outgoing Call for 79821116937 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[14183] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14183] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] DEBUG[14295] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: pushing 0x7f0c300a0130(SIP/zvonobot-00000006) [Aug 18 10:34:11] VERBOSE[14295] bridge_channel.c: Channel SIP/zvonobot-00000006 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:11] DEBUG[14295] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as two channels are required [Aug 18 10:34:11] DEBUG[14295] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14295] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14295] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14295] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14295] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d is already using the new technology. [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1498731843 1498731843 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18334 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:11] DEBUG[14295] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c300a0130(SIP/zvonobot-00000006) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as31e40966 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20bcc5bd [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:11] DEBUG[14295] res_rtp_asterisk.c: (0x7f0c34009e10) RTP changing ssrc from 243567094 to 1798807245 due to a source change [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:11] DEBUG[12894] stasis/app.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' is 2 interested in calls_0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1498731843 1498731843 IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18334 RTP/AVP 0 8 101 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking To) --From tag as31e40966 --To-tag as20bcc5bd [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Stopping retransmission on '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Strict routing enforced for session 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:11] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:11] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:11] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:11] DEBUG[14292] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:11] DEBUG[14292] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14298] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14298] http.c: HTTP Request URI is /ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/record?name=212969_yHqbSyiGmTqhcwsGdMOSOdmkDrfdyHSx&format=wav [Aug 18 10:34:11] DEBUG[14298] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/record] with handler [httpstatus] len 10 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117044@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK405f8c0b Max-Forwards: 70 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:11] DEBUG[14298] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/record] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14298] http.c: match request [ari/bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/record] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14298] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Finding handler for bridges/fa1a4da9-c446-4fa8-95aa-bada67702e1d/record [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[13109] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000001f [Aug 18 10:34:11] DEBUG[13109] stasis/control.c: 212996: Adding to bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 [Aug 18 10:34:11] DEBUG[13109] stasis/app.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Finding handler for fa1a4da9-c446-4fa8-95aa-bada67702e1d [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14298] res_ari.c: No explicit handler found for fa1a4da9-c446-4fa8-95aa-bada67702e1d. Using wildcard bridgeId. [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Finding handler for record [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:11] DEBUG[14298] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:11] DEBUG[14299] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c80056c00(SIP/zvonobot-0000001f) is joining [Aug 18 10:34:11] DEBUG[14298] stasis.c: Creating topic. name: channel:1629282851.315, detail: [Aug 18 10:34:11] DEBUG[14299] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: pushing 0x7f0c80056c00(SIP/zvonobot-0000001f) [Aug 18 10:34:11] DEBUG[14298] stasis.c: Topic 'channel:1629282851.315': 0x7f0c400a40e0 created [Aug 18 10:34:11] VERBOSE[14299] bridge_channel.c: Channel SIP/zvonobot-0000001f joined 'simple_bridge' stasis-bridge <94ef4fc9-246b-4999-9567-b41f8ba44681> [Aug 18 10:34:11] VERBOSE[14297] chan_sip.c: Audio is at 10618 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14298] stasis.c: Creating topic. name: cache:364/channel:1629282851.315, detail: [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Initializing initreq for method INVITE - callid 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116945@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 3 [ 52]: From: ;tag=as09899d91 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 6 [ 60]: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14296] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #126 [Aug 18 10:34:11] DEBUG[14296] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] VERBOSE[14296] dial.c: Called zvonobot/79821116945 [Aug 18 10:34:11] VERBOSE[14297] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] VERBOSE[14297] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[14298] stasis.c: Topic 'cache:364/channel:1629282851.315': 0x7f0c4005a4c0 created [Aug 18 10:34:11] VERBOSE[14297] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1307958254 1307958254 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14926 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09df1e8f;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Initializing initreq for method INVITE - callid 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d114a77 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3d26e887 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116937@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 3 [ 52]: From: ;tag=as17300792 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 6 [ 60]: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14297] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #127 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14297] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] VERBOSE[14297] dial.c: Called zvonobot/79821116937 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] DEBUG[14299] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:11] DEBUG[14299] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[13645] chan_sip.c: Hangup call SIP/zvonobot-00000053, SIP callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[14299] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[13645] res_rtp_asterisk.c: (0x2c67ec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13645] res_rtp_asterisk.c: (0x2c67ec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13645] channel.c: Channel 0x2c6e150 'SIP/zvonobot-00000053' destroying [Aug 18 10:34:11] DEBUG[14299] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.316, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.316': 0x7f0c3007f570 created [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:365/channel:1629282851.316, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:365/channel:1629282851.316': 0x7f0c300b2f10 created [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213051': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213051' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[14299] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14164] channel.c: Channel 0x7f0c38082e90 'SIP/zvonobot-00000087' allocated [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:365/channel:1629282851.316, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:365/channel:1629282851.316': 0x7f0c300b2f10 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.316, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.316': 0x7f0c3007f570 destroyed [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:58', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000053', '', 'AppDial2', '(Outgoing Line)', 10, 0, 'NO ANSWER', 3, '', '213051', '')] [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14164] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[13645] stasis.c: Destroying topic. name: cache:194/channel:213051, detail: [Aug 18 10:34:11] DEBUG[13645] stasis.c: Topic 'cache:194/channel:213051': 0x2c3de20 destroyed [Aug 18 10:34:11] DEBUG[13645] stasis.c: Destroying topic. name: channel:213051, detail: [Aug 18 10:34:11] DEBUG[13645] stasis.c: Topic 'channel:213051': 0x2c44f10 destroyed [Aug 18 10:34:11] DEBUG[14302] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14302] http.c: HTTP Request URI is /ari/channels/213051 [Aug 18 10:34:11] DEBUG[14302] http.c: match request [ari/channels/213051] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14302] http.c: match request [ari/channels/213051] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14302] http.c: match request [ari/channels/213051] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14302] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Finding handler for channels/213051 [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:11] DEBUG[14299] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 is already using the new technology. [Aug 18 10:34:11] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Finding handler for 213051 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:11] DEBUG[14164] res_stasis.c: calls_0: Subscribing to 213094 [Aug 18 10:34:11] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking channels create: Didn't match 213051 [Aug 18 10:34:11] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14299] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c80056c00(SIP/zvonobot-0000001f) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[14302] res_ari.c: Checking channels externalMedia: Didn't match 213051 [Aug 18 10:34:11] DEBUG[14302] res_ari.c: No explicit handler found for 213051. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:11] DEBUG[14164] stasis/app.c: Channel '213094' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1307958254 1307958254 IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[14164] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14164] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Outgoing Call for 79821116946 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] VERBOSE[14303] chan_sip.c: Audio is at 19712 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:11] VERBOSE[14303] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] VERBOSE[14303] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:11] VERBOSE[14303] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14926 RTP/AVP 0 8 101 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Initializing initreq for method INVITE - callid 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116946@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 (Checking To) --From tag as7d114a77 --To-tag as3d26e887 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 3 [ 52]: From: ;tag=as6230d06d [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[14287] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:11] DEBUG[14299] res_rtp_asterisk.c: (0x7f0c8402e520) RTP changing ssrc from 1691611676 to 1946196747 due to a source change [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 6 [ 60]: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14303] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14287] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9 [Aug 18 10:34:11] DEBUG[13109] stasis/app.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' is 2 interested in calls_0 [Aug 18 10:34:11] DEBUG[14303] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Stopping retransmission on '0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:11] DEBUG[14304] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14304] http.c: HTTP Request URI is /ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/record?name=212996_tETNWgXOdyOJtmtuSqFlpvUqMEjxiyua&format=wav [Aug 18 10:34:11] DEBUG[14304] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/record] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14304] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/record] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Strict routing enforced for session 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14304] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/record] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14304] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Finding handler for bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/record [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:11] VERBOSE[14303] dial.c: Called zvonobot/79821116946 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Finding handler for 94ef4fc9-246b-4999-9567-b41f8ba44681 [Aug 18 10:34:11] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14304] res_ari.c: No explicit handler found for 94ef4fc9-246b-4999-9567-b41f8ba44681. Using wildcard bridgeId. [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Finding handler for record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:11] DEBUG[14304] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14304] stasis.c: Creating topic. name: channel:1629282851.317, detail: [Aug 18 10:34:11] DEBUG[14304] stasis.c: Topic 'channel:1629282851.317': 0x7f0c7801a240 created [Aug 18 10:34:11] DEBUG[14304] stasis.c: Creating topic. name: cache:366/channel:1629282851.317, detail: [Aug 18 10:34:11] DEBUG[14304] stasis.c: Topic 'cache:366/channel:1629282851.317': 0x7f0c78053060 created [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117075@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c22028b Max-Forwards: 70 From: ;tag=as7d114a77 To: ;tag=as3d26e887 Contact: Call-ID: 0b7ed345562ac5d65ea2d622724abebb@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:11] DEBUG[14185] channel.c: Channel 0x7f0c940bd420 'SIP/zvonobot-00000088' allocated [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14185] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (5) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116952@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f Max-Forwards: 70 From: ;tag=as64e6e544 To: Contact: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 132789513 132789513 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15512 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14184] channel.c: Channel 0x7f0c8806dac0 'SIP/zvonobot-0000008a' allocated [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (3) INVITE - 5 [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:11] DEBUG[14184] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[14116] res_rtp_asterisk.c: (0x7f0c2400b7d0) RTCP got report of 76 bytes from 178.62.121.41:14089 [Aug 18 10:34:11] VERBOSE[14116] res_rtp_asterisk.c: 0x7f0c2400e650 -- Strict RTP learning complete - Locking on source address 178.62.121.41:14088 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (5) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116955@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc Max-Forwards: 70 From: ;tag=as40bb47c8 To: Contact: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1277373708 1277373708 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14468 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #116 (4) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #116)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (5) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116956@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809 Max-Forwards: 70 From: ;tag=as4d13c830 To: Contact: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 810353375 810353375 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10010 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14184] res_stasis.c: calls_0: Subscribing to 213100 [Aug 18 10:34:11] DEBUG[14184] stasis/app.c: Channel '213100' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14185] res_stasis.c: calls_0: Subscribing to 213102 [Aug 18 10:34:11] DEBUG[14184] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14184] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14185] stasis/app.c: Channel '213102' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Outgoing Call for 79821116940 [Aug 18 10:34:11] DEBUG[14187] channel.c: Channel 0x7f0c900b6f50 'SIP/zvonobot-00000089' allocated [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14187] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] VERBOSE[14306] chan_sip.c: Audio is at 11552 [Aug 18 10:34:11] DEBUG[14185] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14185] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14187] res_stasis.c: calls_0: Subscribing to 213101 [Aug 18 10:34:11] DEBUG[14187] stasis/app.c: Channel '213101' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Outgoing Call for 79821116939 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[14187] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] VERBOSE[14306] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] VERBOSE[14306] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] VERBOSE[14306] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] DEBUG[14187] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14156] app.c: One waitfor failed, trying another [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Outgoing Call for 79821116938 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Initializing initreq for method INVITE - callid 3782ef707142714164cf352b663534ff@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116940@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 3 [ 52]: From: ;tag=as0f1a808c [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 6 [ 60]: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14306] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #83 [Aug 18 10:34:11] DEBUG[14306] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] VERBOSE[14307] chan_sip.c: Audio is at 14986 [Aug 18 10:34:11] VERBOSE[14306] dial.c: Called zvonobot/79821116940 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 From: ;tag=as697b28a1 To: ;tag=as7ad66467 Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" Content-Length: 0 <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:11] VERBOSE[14308] chan_sip.c: Audio is at 16540 [Aug 18 10:34:11] VERBOSE[14308] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 [Aug 18 10:34:11] VERBOSE[14307] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] VERBOSE[14307] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] VERBOSE[14308] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] VERBOSE[14308] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] VERBOSE[14307] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Initializing initreq for method INVITE - callid 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116938@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Initializing initreq for method INVITE - callid 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116939@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as697b28a1 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ad66467 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 3 [ 52]: From: ;tag=as7ed89ca7 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 3 [ 52]: From: ;tag=as2e1ef431 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 6 [ 60]: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 6 [ 60]: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14307] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:34:11] DEBUG[14307] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 (Checking To) --From tag as697b28a1 --To-tag as7ad66467 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[14308] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #94 [Aug 18 10:34:11] DEBUG[14308] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (3) INVITE - 5 [Aug 18 10:34:11] VERBOSE[14307] dial.c: Called zvonobot/79821116938 [Aug 18 10:34:11] DEBUG[13741] app.c: One waitfor failed, trying another [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:11] DEBUG[14111] channel.c: Channel 0x7f0c3c102be0 'Recorder/ARI-0000002c;2' allocated [Aug 18 10:34:11] DEBUG[14111] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[14120] channel.c: Channel 0x7f0c80042760 'Recorder/ARI-0000002e;2' allocated [Aug 18 10:34:11] DEBUG[14120] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[13648] chan_sip.c: Hangup call SIP/zvonobot-00000054, SIP callid 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13072] res_rtp_asterisk.c: (0x7f0c18009d50) DTLS stop [Aug 18 10:34:11] VERBOSE[14308] dial.c: Called zvonobot/79821116939 [Aug 18 10:34:11] DEBUG[13648] res_rtp_asterisk.c: (0x7f0cb00e8f80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13648] res_rtp_asterisk.c: (0x7f0cb00e8f80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13648] channel.c: Channel 0x7f0cb00de970 'SIP/zvonobot-00000054' destroying [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[13644] chan_sip.c: Hangup call SIP/zvonobot-00000052, SIP callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[13996] channel.c: Channel 0x7f0c2001ad30 'Announcer/ARI-00000031;1' allocated [Aug 18 10:34:11] DEBUG[13996] stasis.c: Creating topic. name: channel:1629282851.318, detail: [Aug 18 10:34:11] DEBUG[13996] stasis.c: Topic 'channel:1629282851.318': 0x7f0c20076450 created [Aug 18 10:34:11] DEBUG[13996] stasis.c: Creating topic. name: cache:367/channel:1629282851.318, detail: [Aug 18 10:34:11] DEBUG[13996] stasis.c: Topic 'cache:367/channel:1629282851.318': 0x7f0c20061220 created [Aug 18 10:34:11] DEBUG[14118] channel.c: Channel 0x7f0c780ae220 'Recorder/ARI-0000002d;2' allocated [Aug 18 10:34:11] DEBUG[13644] res_rtp_asterisk.c: (0x7f0c2c0862c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13644] res_rtp_asterisk.c: (0x7f0c2c0862c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13644] channel.c: Channel 0x7f0c2c090fb0 'SIP/zvonobot-00000052' destroying [Aug 18 10:34:11] DEBUG[13072] res_rtp_asterisk.c: (0x7f0c18009d50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13072] res_rtp_asterisk.c: (0x7f0c18009d50) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[13885] chan_sip.c: Hangup call SIP/zvonobot-00000065, SIP callid 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13887] chan_sip.c: Hangup call SIP/zvonobot-00000064, SIP callid 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13072] res_rtp_asterisk.c: (0x7f0c18009d50) ICE stopped [Aug 18 10:34:11] DEBUG[13887] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:11] DEBUG[13885] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:11] DEBUG[13885] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13885] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13885] channel.c: Channel 0x7f0cac032dc0 'SIP/zvonobot-00000065' destroying [Aug 18 10:34:11] DEBUG[13944] channel.c: Channel 0x7f0c280acfd0 'Announcer/ARI-00000026;2' destroying [Aug 18 10:34:11] DEBUG[13072] rtp_engine.c: Destroyed RTP instance '0x7f0c18009d50' [Aug 18 10:34:11] DEBUG[13072] channel.c: Channel 0x7f0c18079740 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50' destroying [Aug 18 10:34:11] DEBUG[13887] res_rtp_asterisk.c: (0x7f0c100f6c90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13887] res_rtp_asterisk.c: (0x7f0c100f6c90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13887] channel.c: Channel 0x7f0c10040760 'SIP/zvonobot-00000064' destroying [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.319, detail: [Aug 18 10:34:11] DEBUG[14118] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[14310] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c80044670(Recorder/ARI-0000002e;2) is joining [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.319': 0x7f0c3007f570 created [Aug 18 10:34:11] DEBUG[14311] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is joining [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:368/channel:1629282851.319, detail: [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6247ms with no response [Aug 18 10:34:11] DEBUG[14313] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c780a94c0(Recorder/ARI-0000002d;2) is joining [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:368/channel:1629282851.319': 0x7f0c300b2f10 created [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[14180] channel.c: Channel 0x7f0c8413e2c0 'SIP/zvonobot-0000008b' allocated [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:11] DEBUG[14180] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Hanging up call 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:11] DEBUG[14313] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: pushing 0x7f0c780a94c0(Recorder/ARI-0000002d;2) [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13550] res_rtp_asterisk.c: (0x7f0c9c008a30) RTCP got report of 100 bytes from 178.62.121.41:15419 [Aug 18 10:34:11] DEBUG[14311] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: pushing 0x7f0c3c10a240(Recorder/ARI-0000002c;2) [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213053': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213053' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:193/channel:213053, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:193/channel:213053': 0x7f0c2c093ab0 destroyed [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213066': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213066' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:256/channel:213066, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:256/channel:213066': 0x7f0cac0320a0 destroyed [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213066, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213066': 0x7f0cac044a30 destroyed [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213064': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213064' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:255/channel:213064, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:255/channel:213064': 0x7f0c100723a0 destroyed [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel 'robot_212964': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel 'robot_212964' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:368/channel:1629282851.319, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:368/channel:1629282851.319': 0x7f0c300b2f10 destroyed [Aug 18 10:34:11] DEBUG[13944] stasis.c: Destroying topic. name: cache:267/channel:1629282842.227, detail: [Aug 18 10:34:11] DEBUG[13944] stasis.c: Topic 'cache:267/channel:1629282842.227': 0x7f0c280ec8f0 destroyed [Aug 18 10:34:11] DEBUG[13944] stasis.c: Destroying topic. name: channel:1629282842.227, detail: [Aug 18 10:34:11] DEBUG[13944] stasis.c: Topic 'channel:1629282842.227': 0x7f0c280f4310 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.319, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.319': 0x7f0c3007f570 destroyed [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:58', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000052', '', 'AppDial2', '(Outgoing Line)', 10, 0, 'NO ANSWER', 3, '', '213053', '')] [Aug 18 10:34:11] DEBUG[14315] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14316] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14316] http.c: HTTP Request URI is /ari/channels/213066 [Aug 18 10:34:11] DEBUG[14317] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.320, detail: [Aug 18 10:34:11] DEBUG[14315] http.c: HTTP Request URI is /ari/channels/213053 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.320': 0x7f0c3007f570 created [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '16541045053beef31ed8e5361337aa22@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[14311] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:44/channel:robot_212964, detail: [Aug 18 10:34:11] DEBUG[14313] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:369/channel:1629282851.320, detail: [Aug 18 10:34:11] DEBUG[13994] channel.c: Channel 0x7f0c2c0c8bd0 'SIP/zvonobot-0000006a' hanging up. Refs: 2 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:44/channel:robot_212964': 0x7f0c1807c280 destroyed [Aug 18 10:34:11] DEBUG[14317] http.c: HTTP Request URI is /ari/channels/213064 [Aug 18 10:34:11] DEBUG[14315] http.c: match request [ari/channels/213053] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14316] http.c: match request [ari/channels/213066] with handler [httpstatus] len 10 [Aug 18 10:34:11] VERBOSE[14313] bridge_channel.c: Channel Recorder/ARI-0000002d;2 joined 'simple_bridge' stasis-bridge <3fc9ee09-2746-49ab-833c-6c9b37b1bb83> [Aug 18 10:34:11] DEBUG[14316] http.c: match request [ari/channels/213066] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14315] http.c: match request [ari/channels/213053] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:369/channel:1629282851.320': 0x7f0c300b2f10 created [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x2c67ec0) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x2c67ec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x2c67ec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x2c67ec0) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x2c67ec0' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[14316] http.c: match request [ari/channels/213066] with handler [ari] len 3 [Aug 18 10:34:11] VERBOSE[14311] bridge_channel.c: Channel Recorder/ARI-0000002c;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:11] DEBUG[14317] http.c: match request [ari/channels/213064] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:369/channel:1629282851.320, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:369/channel:1629282851.320': 0x7f0c300b2f10 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.320, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.320': 0x7f0c3007f570 destroyed [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:11] DEBUG[14315] http.c: match request [ari/channels/213053] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213064, detail: [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14315] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213064': 0x7f0c10041cc0 destroyed [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000065', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213066', '')] [Aug 18 10:34:11] DEBUG[14180] res_stasis.c: calls_0: Subscribing to 213099 [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Finding handler for channels/213053 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212964, detail: [Aug 18 10:34:11] DEBUG[14316] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14310] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pushing 0x7f0c80044670(Recorder/ARI-0000002e;2) [Aug 18 10:34:11] DEBUG[14180] stasis/app.c: Channel '213099' is 1 interested in calls_0 [Aug 18 10:34:11] DEBUG[14180] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:11] DEBUG[14208] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:11] DEBUG[13890] chan_sip.c: Hangup call SIP/zvonobot-00000066, SIP callid 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13890] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:11] DEBUG[13890] res_rtp_asterisk.c: (0x7f0c1c123480) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13890] res_rtp_asterisk.c: (0x7f0c1c123480) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13890] channel.c: Channel 0x7f0c1c12fa80 'SIP/zvonobot-00000066' destroying [Aug 18 10:34:11] DEBUG[13655] chan_sip.c: Hangup call SIP/zvonobot-00000056, SIP callid 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[13655] res_rtp_asterisk.c: (0x7f0c20075860) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13655] res_rtp_asterisk.c: (0x7f0c20075860) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[13655] channel.c: Channel 0x7f0c20081870 'SIP/zvonobot-00000056' destroying [Aug 18 10:34:11] DEBUG[14155] channel.c: Channel 0x7f0c180f2f90 'Recorder/ARI-00000030;2' allocated [Aug 18 10:34:11] DEBUG[14155] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] DEBUG[14317] http.c: match request [ari/channels/213064] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14317] http.c: match request [ari/channels/213064] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14317] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Finding handler for channels/213064 [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Finding handler for 213064 [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking channels create: Didn't match 213064 [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14317] res_ari.c: Checking channels externalMedia: Didn't match 213064 [Aug 18 10:34:11] DEBUG[14317] res_ari.c: No explicit handler found for 213064. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14180] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP got report of 100 bytes from 178.62.121.41:18793 [Aug 18 10:34:11] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP got report of 100 bytes from 178.62.121.41:14675 [Aug 18 10:34:11] DEBUG[14208] http.c: HTTP closing session. Top level [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 221776054 221776054 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10694 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK063aeeb7;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d218141 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6bcc6b47 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 221776054 221776054 IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10694 RTP/AVP 0 8 101 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 (Checking To) --From tag as2d218141 --To-tag as6bcc6b47 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Stopping retransmission on '38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Strict routing enforced for session 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:11] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:11] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117025@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0609deed Max-Forwards: 70 From: ;tag=as2d218141 To: ;tag=as6bcc6b47 Contact: Call-ID: 38a7bfe03e85036a49c051a57967d677@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (3) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6269ms with no response [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #6 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #6)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (1) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (2) INVITE - 5 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '353e057c3fd8d3d27dcbad07571cc18c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb00e8f80) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb00e8f80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb00e8f80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb00e8f80) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb00e8f80' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '63c7430f0c6f8039194559375e330124@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0862c0) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0862c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0862c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0862c0) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c0862c0' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2c5322d560d5755f39711b55002aec77@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS stop [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:robot_212964': 0x7f0c18079310 destroyed [Aug 18 10:34:11] DEBUG[14319] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c180c8c80(Recorder/ARI-00000030;2) is joining [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Outgoing Call for 79821116941 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cac01e130' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f6c90) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f6c90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f6c90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c100f6c90) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c100f6c90' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c123480) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c123480) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c123480) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c123480) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c1c123480' [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '10507dcf059680b46ad884550335c862@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20075860) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20075860) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20075860) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20075860) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c20075860' [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7;received=159.65.48.104 From: ;tag=as3da39e97 To: ;tag=as4a709074 Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3da7eb51" Content-Length: 0 <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK573487d7;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3da39e97 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4a709074 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3da7eb51" [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 (Checking To) --From tag as3da39e97 --To-tag as4a709074 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2596122845f5f4322466678f68967bbf@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Destroying SIP dialog 73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060 [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '73469f9b0059cfd97f7072e02f9ff86f@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS stop [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:11] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9800afc0) ICE RTP transport deallocating [Aug 18 10:34:11] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9800afc0' [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213044': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213044' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:195/channel:213044, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:195/channel:213044': 0x7f0cb00e3b60 destroyed [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213053, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213053': 0x7f0c2c0930c0 destroyed [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213044, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213044': 0x7f0cb00f60a0 destroyed [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:11] DEBUG[14310] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] VERBOSE[14310] bridge_channel.c: Channel Recorder/ARI-0000002e;2 joined 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:11] DEBUG[14310] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c'. Checking compatability for channels 'SIP/zvonobot-00000004' and 'Recorder/ARI-0000002e;2' [Aug 18 10:34:11] DEBUG[14310] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as could not get details [Aug 18 10:34:11] DEBUG[14310] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14310] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14310] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14310] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14310] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:11] DEBUG[14310] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c80044670(Recorder/ARI-0000002e;2) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14311] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa'. Checking compatability for channels 'SIP/zvonobot-00000001' and 'Recorder/ARI-0000002c;2' [Aug 18 10:34:11] DEBUG[14311] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' can not use native RTP bridge as could not get details [Aug 18 10:34:11] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 From: ;tag=as28933467 To: ;tag=as5b70cf89 Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1807314912 1807314912 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58f1977c;received=159.65.48.104 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as28933467 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5b70cf89 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213052': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213052' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:197/channel:213052, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:197/channel:213052': 0x7f0c200840d0 destroyed [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213065': is 0 interested in calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis/app.c: channel '213065' unsubscribed from calls_0 [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: cache:260/channel:213065, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'cache:260/channel:213065': 0x7f0c1c0b08f0 destroyed [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213065, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213065': 0x7f0c1c0575d0 destroyed [Aug 18 10:34:11] DEBUG[20620] stasis.c: Destroying topic. name: channel:213052, detail: [Aug 18 10:34:11] DEBUG[20620] stasis.c: Topic 'channel:213052': 0x7f0c200836d0 destroyed [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Finding handler for channels/213066 [Aug 18 10:34:11] DEBUG[14320] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14320] http.c: HTTP Request URI is /ari/channels/213044 [Aug 18 10:34:11] DEBUG[14320] http.c: match request [ari/channels/213044] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14320] http.c: match request [ari/channels/213044] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14320] http.c: match request [ari/channels/213044] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14320] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Finding handler for channels/213044 [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Finding handler for 213044 [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking channels create: Didn't match 213044 [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14320] res_ari.c: Checking channels externalMedia: Didn't match 213044 [Aug 18 10:34:11] DEBUG[14320] res_ari.c: No explicit handler found for 213044. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14311] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:11] VERBOSE[14318] chan_sip.c: Audio is at 10796 [Aug 18 10:34:11] DEBUG[14311] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Finding handler for 213053 [Aug 18 10:34:11] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14319] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: pushing 0x7f0c180c8c80(Recorder/ARI-00000030;2) [Aug 18 10:34:11] DEBUG[14311] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14311] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14321] http.c: HTTP opening session. Top level [Aug 18 10:34:11] VERBOSE[14318] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking channels create: Didn't match 213053 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14311] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa is already using the new technology. [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:11] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 656, ms is 61 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14311] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:11] VERBOSE[14318] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:11] VERBOSE[14318] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Initializing initreq for method INVITE - callid 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116941@178.62.121.41 SIP/2.0 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 3 [ 52]: From: ;tag=as3f0bc324 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 6 [ 60]: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:11 GMT [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:11] VERBOSE[14318] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #76 [Aug 18 10:34:11] DEBUG[14318] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1807314912 1807314912 IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[14315] res_ari.c: Checking channels externalMedia: Didn't match 213053 [Aug 18 10:34:11] DEBUG[14310] channel.c: Channel Recorder/ARI-0000002e;2 setting read format path: slin -> slin [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel Recorder/ARI-0000002c;2 setting read format path: slin -> slin [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] VERBOSE[14318] dial.c: Called zvonobot/79821116941 [Aug 18 10:34:11] DEBUG[14315] res_ari.c: No explicit handler found for 213053. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14313] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83'. Checking compatability for channels 'SIP/zvonobot-00000033' and 'Recorder/ARI-0000002d;2' [Aug 18 10:34:11] DEBUG[14321] http.c: HTTP Request URI is /ari/channels/213052 [Aug 18 10:34:11] DEBUG[14322] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14319] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:11] VERBOSE[14319] bridge_channel.c: Channel Recorder/ARI-00000030;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:11] DEBUG[14313] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' can not use native RTP bridge as could not get details [Aug 18 10:34:11] DEBUG[14322] http.c: HTTP Request URI is /ari/channels/213065 [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Finding handler for 213066 [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel SIP/zvonobot-00000001 setting write format path: slin -> ulaw [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel SIP/zvonobot-00000001 setting read format path: ulaw -> slin [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel Recorder/ARI-0000002c;2 setting write format path: slin -> slin [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking channels create: Didn't match 213066 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14316] res_ari.c: Checking channels externalMedia: Didn't match 213066 [Aug 18 10:34:11] DEBUG[14316] res_ari.c: No explicit handler found for 213066. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14313] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14313] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14310] channel.c: Channel SIP/zvonobot-00000004 setting write format path: slin -> ulaw [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.321, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.321': 0x7f0c3007f570 created [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:370/channel:1629282851.321, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:370/channel:1629282851.321': 0x7f0c300b2f10 created [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:370/channel:1629282851.321, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:370/channel:1629282851.321': 0x7f0c300b2f10 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.321, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.321': 0x7f0c3007f570 destroyed [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:02', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000064', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213064', '')] [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:11] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:11] DEBUG[14313] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:11] DEBUG[14313] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14313] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 is already using the new technology. [Aug 18 10:34:11] DEBUG[14321] http.c: match request [ari/channels/213052] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14319] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846'. Checking compatability for channels 'SIP/zvonobot-00000038' and 'Recorder/ARI-00000030;2' [Aug 18 10:34:11] DEBUG[14313] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c780a94c0(Recorder/ARI-0000002d;2) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:11] DEBUG[14322] http.c: match request [ari/channels/213065] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14319] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as could not get details [Aug 18 10:34:11] DEBUG[14319] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14169] stasis.c: Creating topic. name: channel:1629282851.322, detail: [Aug 18 10:34:11] DEBUG[14322] http.c: match request [ari/channels/213065] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14321] http.c: match request [ari/channels/213052] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14313] channel.c: Channel Recorder/ARI-0000002d;2 setting read format path: slin -> slin [Aug 18 10:34:11] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14153] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0' [Aug 18 10:34:11] DEBUG[14319] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[13470] channel.c: Channel 0x7f0c88078fc0 'Recorder/ARI-00000014;1' destroying [Aug 18 10:34:11] DEBUG[14319] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[13475] channel.c: Channel 0x7f0c94027db0 'SIP/zvonobot-0000002e' destroying [Aug 18 10:34:11] DEBUG[13475] channel.c: Channel 0x7f0cac05bc90 'Snoop/213012-00000009' destroying [Aug 18 10:34:11] DEBUG[13475] stasis.c: Destroying topic. name: cache:160/channel:1629282835.133, detail: [Aug 18 10:34:11] DEBUG[13475] stasis.c: Topic 'cache:160/channel:1629282835.133': 0x7f0cac05d170 destroyed [Aug 18 10:34:11] DEBUG[13475] stasis.c: Destroying topic. name: channel:1629282835.133, detail: [Aug 18 10:34:11] DEBUG[13475] stasis.c: Topic 'channel:1629282835.133': 0x7f0cac05cc80 destroyed [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14088 RTP/AVP 0 8 101 [Aug 18 10:34:11] DEBUG[14169] stasis.c: Topic 'channel:1629282851.322': 0x7f0c700a3070 created [Aug 18 10:34:11] DEBUG[14321] http.c: match request [ari/channels/213052] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14153] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:11] DEBUG[14153] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14322] http.c: match request [ari/channels/213065] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14169] stasis.c: Creating topic. name: cache:371/channel:1629282851.322, detail: [Aug 18 10:34:11] DEBUG[14169] stasis.c: Topic 'cache:371/channel:1629282851.322': 0x7f0c700a14f0 created [Aug 18 10:34:11] DEBUG[14111] res_stasis_recording.c: 1629282846.265: Sending record(212965_MdQEAXTNGgfRSCaEibwitrXymNroStbk.wav) command [Aug 18 10:34:11] DEBUG[14111] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:11] DEBUG[14111] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14313] channel.c: Channel SIP/zvonobot-00000033 setting write format path: slin -> ulaw [Aug 18 10:34:11] DEBUG[14313] channel.c: Channel SIP/zvonobot-00000033 setting read format path: ulaw -> slin [Aug 18 10:34:11] DEBUG[14313] channel.c: Channel Recorder/ARI-0000002d;2 setting write format path: slin -> slin [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:11] DEBUG[14326] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14324] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14310] channel.c: Channel SIP/zvonobot-00000004 setting read format path: ulaw -> slin [Aug 18 10:34:11] DEBUG[14310] channel.c: Channel Recorder/ARI-0000002e;2 setting write format path: slin -> slin [Aug 18 10:34:11] DEBUG[13541] bridge_channel.c: Setting 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) state from:0 to:1 [Aug 18 10:34:11] DEBUG[14319] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:11] DEBUG[14324] http.c: HTTP Request URI is /ari/channels/213012 [Aug 18 10:34:11] DEBUG[14319] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:11] DEBUG[14324] http.c: match request [ari/channels/213012] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14324] http.c: match request [ari/channels/213012] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[14319] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c180c8c80(Recorder/ARI-00000030;2) is joining simple_bridge technology [Aug 18 10:34:11] DEBUG[14324] http.c: match request [ari/channels/213012] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14324] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:11] DEBUG[14322] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14326] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:11] DEBUG[14326] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Finding handler for channels/213012 [Aug 18 10:34:11] DEBUG[14321] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[14326] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:11] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14326] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14319] channel.c: Channel Recorder/ARI-00000030;2 setting read format path: slin -> slin [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:11] DEBUG[14326] http.c: Match made with [ari] [Aug 18 10:34:11] DEBUG[13541] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: pulling 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel Recorder/ARI-0000002c;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:11] DEBUG[14311] channel.c: Channel Recorder/ARI-0000002c;2 setting write format path: alaw -> slin [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:11] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:11] VERBOSE[13541] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 left 'simple_bridge' stasis-bridge [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:11] DEBUG[13470] stasis.c: Destroying topic. name: cache:155/channel:1629282835.130, detail: [Aug 18 10:34:11] DEBUG[13470] stasis.c: Topic 'cache:155/channel:1629282835.130': 0x7f0c88080f20 destroyed [Aug 18 10:34:11] DEBUG[13470] stasis.c: Destroying topic. name: channel:1629282835.130, detail: [Aug 18 10:34:11] DEBUG[13470] stasis.c: Topic 'channel:1629282835.130': 0x7f0c88080d50 destroyed [Aug 18 10:34:11] DEBUG[14325] app.c: play_and_record: , /var/spool/asterisk/recording/212965_MdQEAXTNGgfRSCaEibwitrXymNroStbk, 'wav' [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Finding handler for 213012 [Aug 18 10:34:11] DEBUG[14321] res_ari.c: Finding handler for channels/213052 [Aug 18 10:34:11] DEBUG[13541] bridge_channel.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71: 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) is leaving simple_bridge technology [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282851.323, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.323': 0x7f0c3007f570 created [Aug 18 10:34:11] DEBUG[20545] stasis.c: Creating topic. name: cache:372/channel:1629282851.323, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:372/channel:1629282851.323': 0x7f0c300b2f10 created [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: cache:372/channel:1629282851.323, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'cache:372/channel:1629282851.323': 0x7f0c300b2f10 destroyed [Aug 18 10:34:11] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282851.323, detail: [Aug 18 10:34:11] DEBUG[20545] stasis.c: Topic 'channel:1629282851.323': 0x7f0c3007f570 destroyed [Aug 18 10:34:11] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:47', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212964', '')] [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking channels create: Didn't match 213012 [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[14322] res_ari.c: Finding handler for channels/213065 [Aug 18 10:34:11] DEBUG[14319] channel.c: Channel SIP/zvonobot-00000038 setting write format path: slin -> alaw [Aug 18 10:34:11] DEBUG[14319] channel.c: Channel SIP/zvonobot-00000038 setting read format path: alaw -> slin [Aug 18 10:34:11] DEBUG[14319] channel.c: Channel Recorder/ARI-00000030;2 setting write format path: slin -> slin [Aug 18 10:34:11] DEBUG[14324] res_ari.c: Checking channels externalMedia: Didn't match 213012 [Aug 18 10:34:11] DEBUG[14325] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Finding handler for bridges [Aug 18 10:34:11] DEBUG[14324] res_ari.c: No explicit handler found for 213012. Using wildcard channelId. [Aug 18 10:34:11] DEBUG[14118] res_stasis_recording.c: 1629282847.266: Sending record(213015_dBotEnLtTVjDPxFVWfqmvQfKraUbAFtr.wav) command [Aug 18 10:34:11] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:11] DEBUG[14118] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:11] DEBUG[14118] http.c: HTTP closing session. Top level [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:11] DEBUG[14321] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[14321] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:11] DEBUG[14326] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:11] DEBUG[14326] stasis.c: Creating topic. name: bridge:cb82e822-34ce-4cab-8b96-97e1b95e246e, detail: [Aug 18 10:34:11] DEBUG[13541] bridge_native_rtp.c: Bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71' can not use native RTP bridge as two channels are required [Aug 18 10:34:11] DEBUG[13541] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:11] DEBUG[14326] stasis.c: Topic 'bridge:cb82e822-34ce-4cab-8b96-97e1b95e246e': 0x7f0c0805c560 created [Aug 18 10:34:11] DEBUG[14322] res_ari.c: Finding handler for channels [Aug 18 10:34:11] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:11] DEBUG[14328] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[13541] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14327] app.c: play_and_record: , /var/spool/asterisk/recording/213015_dBotEnLtTVjDPxFVWfqmvQfKraUbAFtr, 'wav' [Aug 18 10:34:12] DEBUG[14327] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:12] VERBOSE[14327] app.c: x=0, open writing: /var/spool/asterisk/recording/213015_dBotEnLtTVjDPxFVWfqmvQfKraUbAFtr format: wav, 0x7f0c1c149da0 [Aug 18 10:34:12] DEBUG[14328] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:12] DEBUG[13541] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:11] DEBUG[14326] stasis.c: Creating topic. name: cache:373/bridge:cb82e822-34ce-4cab-8b96-97e1b95e246e, detail: [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking To) --From tag as28933467 --To-tag as5b70cf89 [Aug 18 10:34:11] VERBOSE[14325] app.c: x=0, open writing: /var/spool/asterisk/recording/212965_MdQEAXTNGgfRSCaEibwitrXymNroStbk format: wav, 0x7f0c1014bf90 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Stopping retransmission on '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:12] DEBUG[14155] res_stasis_recording.c: 1629282848.273: Sending record(213022_BAYJOCMWmTVQynBoGfsjriMFoVVShhiT.wav) command [Aug 18 10:34:12] DEBUG[14155] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13541] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] DEBUG[13541] bridge.c: Bridge b7adaa29-9b73-48a7-8d8d-8ee58b870f71 is already using the new technology. [Aug 18 10:34:12] DEBUG[14332] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14330] http.c: HTTP opening session. Top level [Aug 18 10:34:11] DEBUG[14321] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14155] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14330] http.c: HTTP Request URI is /ari/channels/213114?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116926&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14326] stasis.c: Topic 'cache:373/bridge:cb82e822-34ce-4cab-8b96-97e1b95e246e': 0x7f0c0805c700 created [Aug 18 10:34:12] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14328] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14332] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:12] DEBUG[14330] http.c: match request [ari/channels/213114] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14330] http.c: match request [ari/channels/213114] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14330] http.c: match request [ari/channels/213114] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14330] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14328] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14222] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:12] DEBUG[14330] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14332] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14222] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Finding handler for channels/213114 [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Finding handler for 213065 [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking channels create: Didn't match 213065 [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14322] res_ari.c: Checking channels externalMedia: Didn't match 213065 [Aug 18 10:34:12] DEBUG[14322] res_ari.c: No explicit handler found for 213065. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:12] DEBUG[14332] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14332] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14313] channel.c: Channel Recorder/ARI-0000002d;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:12] DEBUG[14313] channel.c: Channel Recorder/ARI-0000002d;2 setting write format path: alaw -> slin [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14326] bridge_native_rtp.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' can not use native RTP bridge as two channels are required [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Finding handler for 213114 [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking channels create: Didn't match 213114 [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14330] res_ari.c: Checking channels externalMedia: Didn't match 213114 [Aug 18 10:34:12] DEBUG[14330] res_ari.c: No explicit handler found for 213114. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14326] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:12] DEBUG[14326] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14326] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14332] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:12] DEBUG[14326] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14328] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213012': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[14326] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: calling simple_bridge technology constructor [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[13541] bridge_channel.c: Bridge is returning 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) to write format slin16 [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14326] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: calling simple_bridge technology start [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:12] DEBUG[14336] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213012' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:12] DEBUG[14328] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[13541] channel.c: Channel UnicastRTP/127.0.0.1:50433-0x7f0c080871b0 setting write format path: slin16 -> slin16 [Aug 18 10:34:12] DEBUG[13541] stasis/control.c: robot_213012, b7adaa29-9b73-48a7-8d8d-8ee58b870f71: Channel was departed from bridge [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14336] http.c: HTTP Request URI is /ari/channels/213116?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116924&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.324, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.324': 0x7f0c30145010 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:374/channel:1629282852.324, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:374/channel:1629282852.324': 0x7f0c300ba000 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:374/channel:1629282852.324, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:374/channel:1629282852.324': 0x7f0c300ba000 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.324, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.324': 0x7f0c30145010 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000054', '', 'AppDial2', '(Outgoing Line)', 10, 0, 'NO ANSWER', 3, '', '213044', '')] [Aug 18 10:34:12] DEBUG[14336] http.c: match request [ari/channels/213116] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:12] DEBUG[14336] http.c: match request [ari/channels/213116] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14326] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14336] http.c: match request [ari/channels/213116] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14336] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14336] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:12] DEBUG[14326] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117073@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4d427b67 Max-Forwards: 70 From: ;tag=as28933467 To: ;tag=as5b70cf89 Contact: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #26 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #26)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116954@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8 Max-Forwards: 70 From: ;tag=as2d7c4d21 To: Contact: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 19959187 19959187 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10836 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #105 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #105)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116951@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89 Max-Forwards: 70 From: ;tag=as2ed109a6 To: Contact: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2069655369 2069655369 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18668 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (2) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #6 (2) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #6)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (2) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Finding handler for channels/213116 [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14120] res_stasis_recording.c: 1629282847.267: Sending record(212967_lhjsPNnYXjMyUoluhutJnnhSQwIrUluO.wav) command [Aug 18 10:34:12] DEBUG[14332] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14340] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14339] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14331] app.c: play_and_record: , /var/spool/asterisk/recording/213022_BAYJOCMWmTVQynBoGfsjriMFoVVShhiT, 'wav' [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14340] http.c: HTTP Request URI is /ari/channels/212965/snoop?app=calls_0&spy=in [Aug 18 10:34:12] DEBUG[14340] http.c: match request [ari/channels/212965/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14340] http.c: match request [ari/channels/212965/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14339] http.c: HTTP Request URI is /ari/channels/213115?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116925&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14339] http.c: match request [ari/channels/213115] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14339] http.c: match request [ari/channels/213115] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14339] http.c: match request [ari/channels/213115] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14332] stasis.c: Creating topic. name: bridge:48ea2adf-d916-4f02-8d4c-5c853a6c83c0, detail: [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Finding handler for 213116 [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14340] http.c: match request [ari/channels/212965/snoop] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14342] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking channels create: Didn't match 213116 [Aug 18 10:34:12] DEBUG[14339] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (1) INVITE - 5 [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14342] http.c: HTTP Request URI is /ari/channels/213117?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116923&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14332] stasis.c: Topic 'bridge:48ea2adf-d916-4f02-8d4c-5c853a6c83c0': 0x7f0c2c05fdb0 created [Aug 18 10:34:12] DEBUG[14332] stasis.c: Creating topic. name: cache:375/bridge:48ea2adf-d916-4f02-8d4c-5c853a6c83c0, detail: [Aug 18 10:34:12] DEBUG[14332] stasis.c: Topic 'cache:375/bridge:48ea2adf-d916-4f02-8d4c-5c853a6c83c0': 0x7f0c2c07ace0 created [Aug 18 10:34:12] DEBUG[14332] bridge_native_rtp.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' can not use native RTP bridge as two channels are required [Aug 18 10:34:12] DEBUG[14332] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:12] DEBUG[14332] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14332] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:12] DEBUG[14332] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] DEBUG[14332] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: calling simple_bridge technology constructor [Aug 18 10:34:12] DEBUG[14332] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: calling simple_bridge technology start [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:12] DEBUG[14310] channel.c: Channel Recorder/ARI-0000002e;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Finding handler for 213052 [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking channels create: Didn't match 213052 [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14321] res_ari.c: Checking channels externalMedia: Didn't match 213052 [Aug 18 10:34:12] DEBUG[14321] res_ari.c: No explicit handler found for 213052. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14310] channel.c: Channel Recorder/ARI-0000002e;2 setting write format path: alaw -> slin [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14340] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Finding handler for channels/212965/snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Finding handler for 212965 [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channels create: Didn't match 212965 [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channels externalMedia: Didn't match 212965 [Aug 18 10:34:12] DEBUG[14340] res_ari.c: No explicit handler found for 212965. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Finding handler for snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:12] DEBUG[14340] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:12] DEBUG[14336] res_ari.c: Checking channels externalMedia: Didn't match 213116 [Aug 18 10:34:12] DEBUG[14336] res_ari.c: No explicit handler found for 213116. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006a - start 1629282845.076613 answer 0.000000 end 1629282851.667557 dur 6.590 bill 1629282851.667 dispo NO ANSWER [Aug 18 10:34:12] DEBUG[14339] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14120] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:65/channel:213012, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:65/channel:213012': 0x7f0c940294e0 destroyed [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213012, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213012': 0x7f0c9402ae80 destroyed [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:12] DEBUG[13947] res_rtp_asterisk.c: (0x7f0c340f6d00) RTP audio difference is 784, ms is 69 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[14120] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (6) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116967@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037 Max-Forwards: 70 From: ;tag=as46d7f260 To: Contact: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1625446319 1625446319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18778 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #63 (6) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #63)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116964@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050 Max-Forwards: 70 From: ;tag=as22570a36 To: Contact: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 878955912 878955912 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 From: ;tag=as404b2233 To: ;tag=as0706ba37 Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as404b2233 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0706ba37 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:12] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Finding handler for channels/213115 [Aug 18 10:34:12] DEBUG[13541] stasis/app.c: bridge 'b7adaa29-9b73-48a7-8d8d-8ee58b870f71': is 2 interested in calls_0 [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14332] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Finding handler for 213115 [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking channels create: Didn't match 213115 [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14339] res_ari.c: Checking channels externalMedia: Didn't match 213115 [Aug 18 10:34:12] DEBUG[14339] res_ari.c: No explicit handler found for 213115. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14342] http.c: match request [ari/channels/213117] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14332] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[13541] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:12] DEBUG[14347] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[13497] stasis/control.c: robot_213012: Channel departing bridge [Aug 18 10:34:12] DEBUG[14352] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14350] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14352] http.c: HTTP Request URI is /ari/channels/213118?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116922&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: Allocating new SIP dialog for 030a4fa0531fbde0496fa9846745dd51@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14330] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c240f0400' [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) RTP allocated port 15278 [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE creating session 0.0.0.0:15278 (15278) [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE create [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE add system candidates [Aug 18 10:34:12] DEBUG[14330] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14330] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE add candidate: 159.65.48.104:15278, 2130706431 [Aug 18 10:34:12] DEBUG[14330] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14330] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE add candidate: 10.131.0.10:15278, 2130706431 [Aug 18 10:34:12] DEBUG[14330] rtp_engine.c: RTP instance '0x7f0c240f0400' is setup and ready to go [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) ICE stopped [Aug 18 10:34:12] DEBUG[14330] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14330] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14330] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14330] res_rtp_asterisk.c: (0x7f0c240f0400) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14330] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14330] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14330] chan_sip.c: SIP call-id changed from '030a4fa0531fbde0496fa9846745dd51@127.0.1.1:5060' to '780fccc405c242d348e2e247384adc25@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14330] stasis.c: Creating topic. name: channel:213114, detail: [Aug 18 10:34:12] DEBUG[14330] stasis.c: Topic 'channel:213114': 0x7f0c2413ee20 created [Aug 18 10:34:12] DEBUG[14351] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14330] stasis.c: Creating topic. name: cache:376/channel:213114, detail: [Aug 18 10:34:12] DEBUG[14342] http.c: match request [ari/channels/213117] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 672, ms is 62 [Aug 18 10:34:12] DEBUG[14351] http.c: HTTP Request URI is /ari/channels/213022/snoop?app=calls_0&spy=in [Aug 18 10:34:12] DEBUG[14342] http.c: match request [ari/channels/213117] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14350] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 (Checking To) --From tag as404b2233 --To-tag as0706ba37 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Stopping retransmission on '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117052@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e Max-Forwards: 70 From: ;tag=as404b2233 To: ;tag=as0706ba37 Contact: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116953@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f Max-Forwards: 70 From: ;tag=as42198afd To: Contact: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1894235311 1894235311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (4) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13497] bridge.c: Waiting for 0x7f0c340824a0(UnicastRTP/127.0.0.1:50433-0x7f0c080871b0) bridge thread to die. [Aug 18 10:34:12] DEBUG[13497] stasis/app.c: channel 'robot_213012': is 1 interested in calls_0 [Aug 18 10:34:12] DEBUG[14331] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14352] http.c: match request [ari/channels/213118] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14342] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14350] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14330] stasis.c: Topic 'cache:376/channel:213114': 0x7f0c2413fcb0 created [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14350] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14347] http.c: HTTP Request URI is /ari/channels/213120?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116920&callerId=74950493843 [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14350] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14353] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:12] DEBUG[14328] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:12] DEBUG[14328] stasis.c: Creating topic. name: bridge:ef77827d-b4e8-46c0-9f54-2d75d725926e, detail: [Aug 18 10:34:12] DEBUG[14328] stasis.c: Topic 'bridge:ef77827d-b4e8-46c0-9f54-2d75d725926e': 0x7f0c180babe0 created [Aug 18 10:34:12] DEBUG[14328] stasis.c: Creating topic. name: cache:377/bridge:ef77827d-b4e8-46c0-9f54-2d75d725926e, detail: [Aug 18 10:34:12] DEBUG[14328] stasis.c: Topic 'cache:377/bridge:ef77827d-b4e8-46c0-9f54-2d75d725926e': 0x7f0c180c16a0 created [Aug 18 10:34:12] DEBUG[14352] http.c: match request [ari/channels/213118] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14288] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14351] http.c: match request [ari/channels/213022/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14350] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14342] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14351] http.c: match request [ari/channels/213022/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14351] http.c: match request [ari/channels/213022/snoop] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:12] DEBUG[14350] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:12] DEBUG[14350] stasis.c: Creating topic. name: bridge:6199a092-f834-41fb-9e43-7eb7ef40551d, detail: [Aug 18 10:34:12] DEBUG[14350] stasis.c: Topic 'bridge:6199a092-f834-41fb-9e43-7eb7ef40551d': 0x7f0c740ad530 created [Aug 18 10:34:12] DEBUG[14350] stasis.c: Creating topic. name: cache:378/bridge:6199a092-f834-41fb-9e43-7eb7ef40551d, detail: [Aug 18 10:34:12] DEBUG[14350] stasis.c: Topic 'cache:378/bridge:6199a092-f834-41fb-9e43-7eb7ef40551d': 0x7f0c740adee0 created [Aug 18 10:34:12] VERBOSE[14331] app.c: x=0, open writing: /var/spool/asterisk/recording/213022_BAYJOCMWmTVQynBoGfsjriMFoVVShhiT format: wav, 0x7f0c2001a430 [Aug 18 10:34:12] DEBUG[14353] http.c: HTTP Request URI is /ari/channels/213119?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116921&callerId=74950493843 [Aug 18 10:34:12] DEBUG[13497] channel.c: Channel 0x7f0c08053190 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14354] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14328] bridge_native_rtp.c: Bridge 'ef77827d-b4e8-46c0-9f54-2d75d725926e' can not use native RTP bridge as two channels are required [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[14349] app.c: play_and_record: , /var/spool/asterisk/recording/212967_lhjsPNnYXjMyUoluhutJnnhSQwIrUluO, 'wav' [Aug 18 10:34:12] DEBUG[14349] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:12] DEBUG[14352] http.c: match request [ari/channels/213118] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:12] DEBUG[14354] http.c: HTTP Request URI is /ari/channels/213122?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116918&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14351] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14352] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14350] bridge_native_rtp.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' can not use native RTP bridge as two channels are required [Aug 18 10:34:12] DEBUG[14350] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:12] DEBUG[14350] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14350] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:12] DEBUG[14350] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] DEBUG[14350] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: calling simple_bridge technology constructor [Aug 18 10:34:12] DEBUG[14350] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: calling simple_bridge technology start [Aug 18 10:34:12] DEBUG[14350] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14328] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:12] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 640, ms is 60 [Aug 18 10:34:12] DEBUG[14350] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Finding handler for channels/213022/snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Finding handler for 213022 [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channels create: Didn't match 213022 [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channels externalMedia: Didn't match 213022 [Aug 18 10:34:12] DEBUG[14351] res_ari.c: No explicit handler found for 213022. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Finding handler for snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:12] DEBUG[14351] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14288] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14328] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14328] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:12] DEBUG[14328] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] DEBUG[14328] bridge.c: Bridge ef77827d-b4e8-46c0-9f54-2d75d725926e: calling simple_bridge technology constructor [Aug 18 10:34:12] DEBUG[14328] bridge.c: Bridge ef77827d-b4e8-46c0-9f54-2d75d725926e: calling simple_bridge technology start [Aug 18 10:34:12] DEBUG[14328] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Finding handler for channels/213117 [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14356] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14352] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14354] http.c: match request [ari/channels/213122] with handler [httpstatus] len 10 [Aug 18 10:34:12] VERBOSE[14349] app.c: x=0, open writing: /var/spool/asterisk/recording/212967_lhjsPNnYXjMyUoluhutJnnhSQwIrUluO format: wav, 0x7f0c400a6570 [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[14357] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: Allocating new SIP dialog for 6d2970040a60e27b3d44326664d0fd7a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14336] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280cff50' [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) RTP allocated port 12662 [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE creating session 0.0.0.0:12662 (12662) [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE create [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE add system candidates [Aug 18 10:34:12] DEBUG[14336] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14336] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE add candidate: 159.65.48.104:12662, 2130706431 [Aug 18 10:34:12] DEBUG[14336] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14336] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE add candidate: 10.131.0.10:12662, 2130706431 [Aug 18 10:34:12] DEBUG[14336] rtp_engine.c: RTP instance '0x7f0c280cff50' is setup and ready to go [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) ICE stopped [Aug 18 10:34:12] DEBUG[14336] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14336] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14336] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14336] res_rtp_asterisk.c: (0x7f0c280cff50) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14336] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14347] http.c: match request [ari/channels/213120] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14354] http.c: match request [ari/channels/213122] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[14354] http.c: match request [ari/channels/213122] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14355] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14328] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14354] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14356] http.c: HTTP Request URI is /ari/channels/212967/snoop?app=calls_0&spy=in [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Finding handler for channels/213118 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14354] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Finding handler for 213118 [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking channels create: Didn't match 213118 [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14352] res_ari.c: Checking channels externalMedia: Didn't match 213118 [Aug 18 10:34:12] DEBUG[14352] res_ari.c: No explicit handler found for 213118. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14353] http.c: match request [ari/channels/213119] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14356] http.c: match request [ari/channels/212967/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14353] http.c: match request [ari/channels/213119] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.326, detail: [Aug 18 10:34:12] DEBUG[14336] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[13661] chan_sip.c: Hangup call SIP/zvonobot-00000055, SIP callid 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[13661] res_rtp_asterisk.c: (0x7f0cac0660c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13661] res_rtp_asterisk.c: (0x7f0cac0660c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13661] channel.c: Channel 0x7f0cac079f80 'SIP/zvonobot-00000055' destroying [Aug 18 10:34:12] DEBUG[13939] chan_sip.c: Hangup call SIP/zvonobot-00000067, SIP callid 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14353] http.c: match request [ari/channels/213119] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14347] http.c: match request [ari/channels/213120] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13041] chan_sip.c: Hangup call SIP/zvonobot-0000001a, SIP callid 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Finding handler for channels/213122 [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[13939] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6368ms with no response [Aug 18 10:34:12] DEBUG[13950] chan_sip.c: Hangup call SIP/zvonobot-0000006b, SIP callid 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[13950] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:12] DEBUG[13950] res_rtp_asterisk.c: (0x7f0c0802d370) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13950] res_rtp_asterisk.c: (0x7f0c0802d370) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13950] channel.c: Channel 0x7f0c080ee420 'SIP/zvonobot-0000006b' destroying [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.326': 0x7f0c3007f570 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:379/channel:1629282852.326, detail: [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[13939] res_rtp_asterisk.c: (0x7f0c180cf000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13658] chan_sip.c: Hangup call SIP/zvonobot-00000058, SIP callid 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14336] chan_sip.c: SIP call-id changed from '6d2970040a60e27b3d44326664d0fd7a@127.0.1.1:5060' to '4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14336] stasis.c: Creating topic. name: channel:213116, detail: [Aug 18 10:34:12] DEBUG[14336] stasis.c: Topic 'channel:213116': 0x7f0c280d2550 created [Aug 18 10:34:12] DEBUG[13041] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13041] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13041] channel.c: Channel 0x7f0c78021200 'SIP/zvonobot-0000001a' destroying [Aug 18 10:34:12] DEBUG[14357] http.c: HTTP Request URI is /ari/channels/213015/snoop?app=calls_0&spy=in [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Finding handler for 213117 [Aug 18 10:34:12] DEBUG[14355] http.c: HTTP Request URI is /ari/channels/213121?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116919&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13939] res_rtp_asterisk.c: (0x7f0c180cf000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking channels create: Didn't match 213117 [Aug 18 10:34:12] DEBUG[14356] http.c: match request [ari/channels/212967/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14357] http.c: match request [ari/channels/213015/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14359] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:379/channel:1629282852.326': 0x7f0c300b2f10 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:379/channel:1629282852.326, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:379/channel:1629282852.326': 0x7f0c300b2f10 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.326, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.326': 0x7f0c3007f570 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000056', '', 'AppDial2', '(Outgoing Line)', 10, 0, 'NO ANSWER', 3, '', '213052', '')] [Aug 18 10:34:12] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14342] res_ari.c: Checking channels externalMedia: Didn't match 213117 [Aug 18 10:34:12] DEBUG[14342] res_ari.c: No explicit handler found for 213117. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13658] res_rtp_asterisk.c: (0x7f0c280ef5e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13658] res_rtp_asterisk.c: (0x7f0c280ef5e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13658] channel.c: Channel 0x7f0c28104e60 'SIP/zvonobot-00000058' destroying [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Finding handler for 213122 [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking channels create: Didn't match 213122 [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14354] res_ari.c: Checking channels externalMedia: Didn't match 213122 [Aug 18 10:34:12] DEBUG[14354] res_ari.c: No explicit handler found for 213122. Using wildcard channelId. [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14041] channel.c: Channel 0x7f0ca00ed5f0 'SIP/zvonobot-0000006f' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14355] http.c: match request [ari/channels/213121] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14353] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14353] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14356] http.c: match request [ari/channels/212967/snoop] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14355] http.c: match request [ari/channels/213121] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14355] http.c: match request [ari/channels/213121] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14355] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14355] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: Allocating new SIP dialog for 5de0e37a09642c035a9d01b24a16fa64@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14339] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c340b9d00' [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) RTP allocated port 17556 [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE creating session 0.0.0.0:17556 (17556) [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE create [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE add system candidates [Aug 18 10:34:12] DEBUG[14339] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14339] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE add candidate: 159.65.48.104:17556, 2130706431 [Aug 18 10:34:12] DEBUG[14336] stasis.c: Creating topic. name: cache:380/channel:213116, detail: [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13939] channel.c: Channel 0x7f0c180bb1b0 'SIP/zvonobot-00000067' destroying [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14339] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213067': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213067' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:269/channel:213067, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:269/channel:213067': 0x7f0c0803d410 destroyed [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '212988': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '212988' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:12] DEBUG[14336] stasis.c: Topic 'cache:380/channel:213116': 0x7f0c280ead30 created [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13056] res_rtp_asterisk.c: (0x7f0cb0001f70) RTCP got report of 100 bytes from 178.62.121.41:11671 [Aug 18 10:34:12] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14288] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (2) INVITE - 5 [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.328, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.328': 0x7f0c3013cab0 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:381/channel:1629282852.328, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:381/channel:1629282852.328': 0x7f0c3007f570 created [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6598ms with no response [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14347] http.c: match request [ari/channels/213120] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14339] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE add candidate: 10.131.0.10:17556, 2130706431 [Aug 18 10:34:12] DEBUG[14339] rtp_engine.c: RTP instance '0x7f0c340b9d00' is setup and ready to go [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE stopped [Aug 18 10:34:12] DEBUG[14339] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14339] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14339] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14339] res_rtp_asterisk.c: (0x7f0c340b9d00) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14339] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14339] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14339] chan_sip.c: SIP call-id changed from '5de0e37a09642c035a9d01b24a16fa64@127.0.1.1:5060' to '15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14339] stasis.c: Creating topic. name: channel:213115, detail: [Aug 18 10:34:12] DEBUG[14339] stasis.c: Topic 'channel:213115': 0x7f0c340f1590 created [Aug 18 10:34:12] DEBUG[14339] stasis.c: Creating topic. name: cache:382/channel:213115, detail: [Aug 18 10:34:12] DEBUG[14339] stasis.c: Topic 'cache:382/channel:213115': 0x7f0c340411c0 created [Aug 18 10:34:12] DEBUG[14357] http.c: match request [ari/channels/213015/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Finding handler for channels/213119 [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14035] channel.c: Channel 0x7f0c400470e0 'SIP/zvonobot-00000068' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14357] http.c: match request [ari/channels/213015/snoop] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1200, ms is 95 [Aug 18 10:34:12] DEBUG[14357] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Finding handler for channels/213015/snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14359] http.c: HTTP Request URI is /ari/channels/213123?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116917&callerId=74950493843 [Aug 18 10:34:12] DEBUG[14362] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 768, ms is 68 [Aug 18 10:34:12] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6574ms with no response [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 From: ;tag=as39a2ec19 To: ;tag=as70b1d74e Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1138223289 1138223289 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11310 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b182323;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39a2ec19 [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:12] DEBUG[14039] channel.c: Channel 0x7f0c9409b680 'SIP/zvonobot-0000006e' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Finding handler for 213015 [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channels create: Didn't match 213015 [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channels externalMedia: Didn't match 213015 [Aug 18 10:34:12] DEBUG[14357] res_ari.c: No explicit handler found for 213015. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Finding handler for snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:12] DEBUG[14357] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13041] stasis.c: Destroying topic. name: cache:33/channel:212988, detail: [Aug 18 10:34:12] DEBUG[13041] stasis.c: Topic 'cache:33/channel:212988': 0x7f0c780226d0 destroyed [Aug 18 10:34:12] DEBUG[13041] stasis.c: Destroying topic. name: channel:212988, detail: [Aug 18 10:34:12] DEBUG[13041] stasis.c: Topic 'channel:212988': 0x7f0c78022ac0 destroyed [Aug 18 10:34:12] DEBUG[14362] http.c: HTTP Request URI is /ari/channels/212988 [Aug 18 10:34:12] DEBUG[14347] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14361] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14246] channel.c: Channel 0x7f0c7c0a2360 'SIP/zvonobot-0000008c' allocated [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:12] DEBUG[14246] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:12] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14362] http.c: match request [ari/channels/212988] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14362] http.c: match request [ari/channels/212988] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14362] http.c: match request [ari/channels/212988] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70b1d74e [Aug 18 10:34:12] DEBUG[14272] res_rtp_asterisk.c: (0x7f0c98083570) RTP audio difference is 640, ms is 60 [Aug 18 10:34:12] DEBUG[14359] http.c: match request [ari/channels/213123] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14361] http.c: HTTP Request URI is /ari/channels/213067 [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 736, ms is 66 [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14347] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1138223289 1138223289 IN IP4 178.62.121.41 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11310 RTP/AVP 0 8 101 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 (Checking To) --From tag as39a2ec19 --To-tag as70b1d74e [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Stopping retransmission on '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Strict routing enforced for session 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:12] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:12] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117071@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4fa57cec Max-Forwards: 70 From: ;tag=as39a2ec19 To: ;tag=as70b1d74e Contact: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #6 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #6)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (4) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116947@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64 Max-Forwards: 70 From: ;tag=as293a990c To: Contact: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 959406650 959406650 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10588 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (6) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116966@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea Max-Forwards: 70 From: ;tag=as7eb98fd0 To: Contact: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 523501849 523501849 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10288 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Destroying SIP dialog 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0660c0) DTLS stop [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0660c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0660c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0660c0) ICE RTP transport deallocating [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cac0660c0' [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Destroying SIP dialog 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0802d370) DTLS stop [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0802d370) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0802d370) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0802d370) ICE RTP transport deallocating [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c0802d370' [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS stop [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c780068b0) ICE RTP transport deallocating [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c780068b0' [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Destroying SIP dialog 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280ef5e0) DTLS stop [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280ef5e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280ef5e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280ef5e0) ICE RTP transport deallocating [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c280ef5e0' [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Destroying SIP dialog 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180cf000) DTLS stop [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180cf000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180cf000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180cf000) ICE RTP transport deallocating [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c180cf000' [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (6) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116962@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8 Max-Forwards: 70 From: ;tag=as0a05f417 To: Contact: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1587432262 1587432262 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10498 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (4) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6358ms with no response [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Finding handler for 213119 [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking channels create: Didn't match 213119 [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14353] res_ari.c: Checking channels externalMedia: Didn't match 213119 [Aug 18 10:34:12] DEBUG[14353] res_ari.c: No explicit handler found for 213119. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14361] http.c: match request [ari/channels/213067] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14288] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14362] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:381/channel:1629282852.328, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:381/channel:1629282852.328': 0x7f0c3007f570 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.328, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.328': 0x7f0c3013cab0 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000066', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213065', '')] [Aug 18 10:34:12] DEBUG[14356] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Finding handler for channels/213121 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Finding handler for channels/213120 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213046': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213046' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:196/channel:213046, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:196/channel:213046': 0x7f0cac07c750 destroyed [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213049': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213049' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:200/channel:213049, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:200/channel:213049': 0x7f0c281079f0 destroyed [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: Allocating new SIP dialog for 6c4a39fb1993bd4d5c79ebe15572759c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: Allocating new SIP dialog for 03501d1610513f6e5efc9b1b2d1380a4@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14361] http.c: match request [ari/channels/213067] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 864, ms is 74 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14342] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c11fbd0' [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) RTP allocated port 18958 [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE creating session 0.0.0.0:18958 (18958) [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE create [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE add system candidates [Aug 18 10:34:12] DEBUG[14342] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14342] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE add candidate: 159.65.48.104:18958, 2130706431 [Aug 18 10:34:12] DEBUG[14342] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14342] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE add candidate: 10.131.0.10:18958, 2130706431 [Aug 18 10:34:12] DEBUG[14342] rtp_engine.c: RTP instance '0x7f0c3c11fbd0' is setup and ready to go [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE stopped [Aug 18 10:34:12] DEBUG[14342] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14342] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14342] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14342] res_rtp_asterisk.c: (0x7f0c3c11fbd0) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14342] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14342] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14342] chan_sip.c: SIP call-id changed from '6c4a39fb1993bd4d5c79ebe15572759c@127.0.1.1:5060' to '5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14342] stasis.c: Creating topic. name: channel:213117, detail: [Aug 18 10:34:12] DEBUG[14342] stasis.c: Topic 'channel:213117': 0x7f0c3c118690 created [Aug 18 10:34:12] DEBUG[14342] stasis.c: Creating topic. name: cache:383/channel:213117, detail: [Aug 18 10:34:12] DEBUG[14342] stasis.c: Topic 'cache:383/channel:213117': 0x7f0c3c133fd0 created [Aug 18 10:34:12] DEBUG[14359] http.c: match request [ari/channels/213123] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Finding handler for channels/212988 [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Finding handler for 212988 [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking channels create: Didn't match 212988 [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14362] res_ari.c: Checking channels externalMedia: Didn't match 212988 [Aug 18 10:34:12] DEBUG[14362] res_ari.c: No explicit handler found for 212988. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14361] http.c: match request [ari/channels/213067] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14361] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14368] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[14359] http.c: match request [ari/channels/213123] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14367] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14359] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTP audio difference is 880, ms is 130 [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: Allocating new SIP dialog for 52d33b4e1ef50ce26ebcab9e3f9d5a77@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14352] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c0c2520' [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) RTP allocated port 12284 [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE creating session 0.0.0.0:12284 (12284) [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE create [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE add system candidates [Aug 18 10:34:12] DEBUG[14352] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14352] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE add candidate: 159.65.48.104:12284, 2130706431 [Aug 18 10:34:12] DEBUG[14352] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14352] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE add candidate: 10.131.0.10:12284, 2130706431 [Aug 18 10:34:12] DEBUG[14352] rtp_engine.c: RTP instance '0x7f0c7c0c2520' is setup and ready to go [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE stopped [Aug 18 10:34:12] DEBUG[14352] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14352] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14352] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14352] res_rtp_asterisk.c: (0x7f0c7c0c2520) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14352] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14352] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14352] chan_sip.c: SIP call-id changed from '52d33b4e1ef50ce26ebcab9e3f9d5a77@127.0.1.1:5060' to '4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14352] stasis.c: Creating topic. name: channel:213118, detail: [Aug 18 10:34:12] DEBUG[14352] stasis.c: Topic 'channel:213118': 0x7f0c7c094c80 created [Aug 18 10:34:12] DEBUG[14352] stasis.c: Creating topic. name: cache:384/channel:213118, detail: [Aug 18 10:34:12] DEBUG[14352] stasis.c: Topic 'cache:384/channel:213118': 0x7f0c7c0a3370 created [Aug 18 10:34:12] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14226] channel.c: Channel 0x7f0c3c111a10 'SIP/zvonobot-0000008d' allocated [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:12] DEBUG[14226] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:12] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Finding handler for channels/213067 [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14048] channel.c: Channel 0x7f0c8c050630 'SIP/zvonobot-00000071' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14354] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c84147390' [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) RTP allocated port 12398 [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE creating session 0.0.0.0:12398 (12398) [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE create [Aug 18 10:34:12] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP audio difference is 664, ms is 103 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 992, ms is 82 [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP audio difference is 696, ms is 107 [Aug 18 10:34:12] DEBUG[14367] http.c: HTTP Request URI is /ari/channels/213046 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.332, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.332': 0x7f0c3013e980 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:385/channel:1629282852.332, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:385/channel:1629282852.332': 0x7f0c30122290 created [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213067, detail: [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14113] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4;received=159.65.48.104 From: ;tag=as08a5ad00 To: ;tag=as331133ce Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27a12182" Content-Length: 0 <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08a5ad00 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as331133ce [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27a12182" [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:12] DEBUG[14154] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE add system candidates [Aug 18 10:34:12] DEBUG[14354] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14354] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Finding handler for 213120 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213067': 0x7f0c0806bd50 destroyed [Aug 18 10:34:12] DEBUG[14368] http.c: HTTP Request URI is /ari/channels/213049 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking channels create: Didn't match 213120 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14347] res_ari.c: Checking channels externalMedia: Didn't match 213120 [Aug 18 10:34:12] DEBUG[14359] http.c: HTTP consuming request body [Aug 18 10:34:12] DEBUG[14368] http.c: match request [ari/channels/213049] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14246] res_stasis.c: calls_0: Subscribing to 213106 [Aug 18 10:34:12] DEBUG[14246] stasis/app.c: Channel '213106' is 1 interested in calls_0 [Aug 18 10:34:12] DEBUG[14368] http.c: match request [ari/channels/213049] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14246] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14246] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Finding handler for 213067 [Aug 18 10:34:12] DEBUG[14347] res_ari.c: No explicit handler found for 213120. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Outgoing Call for 79821116934 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking channels create: Didn't match 213067 [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13347] audiohook.c: Audiohook 0x7f0c74025f00 has stale audio in its factories. Flushing them both [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 (Checking To) --From tag as08a5ad00 --To-tag as331133ce [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE add candidate: 159.65.48.104:12398, 2130706431 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6461ms with no response [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP audio difference is 680, ms is 105 [Aug 18 10:34:12] DEBUG[14361] res_ari.c: Checking channels externalMedia: Didn't match 213067 [Aug 18 10:34:12] DEBUG[14361] res_ari.c: No explicit handler found for 213067. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14367] http.c: match request [ari/channels/213046] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Finding handler for channels/213123 [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Finding handler for channels/212967/snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14367] http.c: match request [ari/channels/213046] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14050] channel.c: Channel 0x7f0c9803da40 'SIP/zvonobot-00000070' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:385/channel:1629282852.332, detail: [Aug 18 10:34:12] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:385/channel:1629282852.332': 0x7f0c30122290 destroyed [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213068': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.332, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.332': 0x7f0c3013e980 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000002e', '', 'Stasis', 'calls_0', 17, 12, 'ANSWERED', 3, '', '213012', '')] [Aug 18 10:34:12] DEBUG[14354] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14354] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE add candidate: 10.131.0.10:12398, 2130706431 [Aug 18 10:34:12] DEBUG[14354] rtp_engine.c: RTP instance '0x7f0c84147390' is setup and ready to go [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) ICE stopped [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Finding handler for 212967 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channels create: Didn't match 212967 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channels externalMedia: Didn't match 212967 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: No explicit handler found for 212967. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Finding handler for snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14376] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:12] DEBUG[14367] http.c: match request [ari/channels/213046] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14226] res_stasis.c: calls_0: Subscribing to 213104 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213068' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:261/channel:213068, detail: [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:261/channel:213068': 0x7f0c180c9110 destroyed [Aug 18 10:34:12] DEBUG[14376] http.c: HTTP Request URI is /ari/channels/213068 [Aug 18 10:34:12] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14368] http.c: match request [ari/channels/213049] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14367] http.c: Match made with [ari] [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (6) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116965@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32d1309a Max-Forwards: 70 From: ;tag=as44c869a1 To: Contact: Call-ID: 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 398643289 398643289 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Finding handler for 213123 [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking channels create: Didn't match 213123 [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14359] res_ari.c: Checking channels externalMedia: Didn't match 213123 [Aug 18 10:34:12] DEBUG[14359] res_ari.c: No explicit handler found for 213123. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14354] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14226] stasis/app.c: Channel '213104' is 1 interested in calls_0 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:12] DEBUG[14226] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14226] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14368] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 From: ;tag=as366f0ed0 To: ;tag=as0a7e373f Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22f7b87b;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as366f0ed0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0a7e373f [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[14376] http.c: match request [ari/channels/213068] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14376] http.c: match request [ari/channels/213068] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.333, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.333': 0x7f0c3013e980 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:386/channel:1629282852.333, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:386/channel:1629282852.333': 0x7f0c30122290 created [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14376] http.c: match request [ari/channels/213068] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Finding handler for channels/213046 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:12] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 (Checking To) --From tag as366f0ed0 --To-tag as0a7e373f [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 636b9cd14097056946c6db972ebebc4b@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (3) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6412ms with no response [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14376] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Finding handler for channels/213068 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:12] DEBUG[14354] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Finding handler for 213068 [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking channels create: Didn't match 213068 [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14376] res_ari.c: Checking channels externalMedia: Didn't match 213068 [Aug 18 10:34:12] DEBUG[14376] res_ari.c: No explicit handler found for 213068. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213046, detail: [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 656, ms is 61 [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213046': 0x7f0cac07bcd0 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:386/channel:1629282852.333, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:386/channel:1629282852.333': 0x7f0c30122290 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.333, detail: [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Finding handler for 213046 [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking channels create: Didn't match 213046 [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] WARNING[14099] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000027;1 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Finding handler for 213121 [Aug 18 10:34:12] DEBUG[14051] channel.c: Channel 0x7f0ca80e0110 'SIP/zvonobot-00000072' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Outgoing Call for 79821116936 [Aug 18 10:34:12] DEBUG[14367] res_ari.c: Checking channels externalMedia: Didn't match 213046 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking channels create: Didn't match 213121 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Finding handler for channels/213049 [Aug 18 10:34:12] DEBUG[14367] res_ari.c: No explicit handler found for 213046. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14354] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14354] res_rtp_asterisk.c: (0x7f0c84147390) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14354] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14354] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Finding handler for channels [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.333': 0x7f0c3013e980 destroyed [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 From: ;tag=as76ba9cfd To: ;tag=as041dcfd6 Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" Content-Length: 0 <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as76ba9cfd [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as041dcfd6 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 (Checking To) --From tag as76ba9cfd --To-tag as041dcfd6 [Aug 18 10:34:12] DEBUG[14354] chan_sip.c: SIP call-id changed from '03501d1610513f6e5efc9b1b2d1380a4@127.0.1.1:5060' to '7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14354] stasis.c: Creating topic. name: channel:213122, detail: [Aug 18 10:34:12] DEBUG[14354] stasis.c: Topic 'channel:213122': 0x7f0c84109ae0 created [Aug 18 10:34:12] DEBUG[14354] stasis.c: Creating topic. name: cache:387/channel:213122, detail: [Aug 18 10:34:12] DEBUG[14354] stasis.c: Topic 'cache:387/channel:213122': 0x7f0c84096460 created [Aug 18 10:34:12] DEBUG[14355] res_ari.c: Checking channels externalMedia: Didn't match 213121 [Aug 18 10:34:12] DEBUG[14355] res_ari.c: No explicit handler found for 213121. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #65 (5) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #65)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116948@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6 Max-Forwards: 70 From: ;tag=as0cd290ec To: Contact: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1313767596 1313767596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11608 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2;received=159.65.48.104 From: ;tag=as00c25c39 To: ;tag=as1eec528d Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4edcab50" Content-Length: 0 <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42d5b1e2;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as00c25c39 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1eec528d [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4edcab50" [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 (Checking To) --From tag as00c25c39 --To-tag as1eec528d [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1ee655842d2ed684574010b3091c860a@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6b879d2d;received=159.65.48.104 From: ;tag=as410f495a To: ;tag=as5cb94950 Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 559217357 559217357 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 11140 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6b879d2d;received=159.65.48.104 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as410f495a [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5cb94950 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 559217357 559217357 IN IP4 178.62.121.41 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 11140 RTP/AVP 0 8 101 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: = Looking for Call ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 (Checking To) --From tag as410f495a --To-tag as5cb94950 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Stopping retransmission on '51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:12] DEBUG[14356] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:55', '"" <>', '', 's', 'default', 'Snoop/213012-00000009', 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0', 'Stasis', 'calls_0', 14, 14, 'ANSWERED', 3, '', '1629282835.133', '')] [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[14347] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] VERBOSE[14372] chan_sip.c: Audio is at 19144 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] VERBOSE[14372] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:12] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Got SDP version 559217357 and unique parts [root 559217357 IN IP4 178.62.121.41] [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 559217357 559217357 IN IP4 178.62.121.41... OK. [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:12] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:12] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800c760) ICE set role failed; no ice instance [Aug 18 10:34:12] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800c760) RTCP setting address on RTP instance [Aug 18 10:34:12] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c7800dea0 -- Strict RTP learning after remote address set to: 178.62.121.41:11140 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:11140 [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb005bd98) from 0x7f0c147e2330 to 0x7f0c7800c938 [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0086c58) from 0x7f0c147e2330 to 0x7f0c7800c938 [Aug 18 10:34:12] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0087df8) from 0x7f0c147e2330 to 0x7f0c7800c938 [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800c760) RTCP ignoring duplicate property [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:12] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000a setting read format path: alaw -> alaw [Aug 18 10:34:12] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000a setting write format path: alaw -> alaw [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800c760) DTLS - ast_rtp_activate rtp=0x7f0c7800dea0 - setup and perform DTLS' [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800dea0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:12] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7800dea0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:12] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:12] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:12] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Strict routing enforced for session 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213049, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213049': 0x7f0c28106f70 destroyed [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213068, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213068': 0x7f0c180a9610 destroyed [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: Allocating new SIP dialog for 2b8ca4bb22e45d7255b22df85bd105fb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14353] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7804d100' [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) RTP allocated port 15158 [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE creating session 0.0.0.0:15158 (15158) [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE create [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE add system candidates [Aug 18 10:34:12] DEBUG[14353] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14353] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE add candidate: 159.65.48.104:15158, 2130706431 [Aug 18 10:34:12] DEBUG[14347] chan_sip.c: Allocating new SIP dialog for 696f5d62547db27838918a632a95bd69@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:12] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:12] VERBOSE[14377] chan_sip.c: Audio is at 15928 [Aug 18 10:34:12] VERBOSE[14377] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:12] VERBOSE[14377] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:12] VERBOSE[14377] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Initializing initreq for method INVITE - callid 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116936@178.62.121.41 SIP/2.0 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 3 [ 52]: From: ;tag=as3ee6d51f [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 6 [ 60]: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:12 GMT [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:12] VERBOSE[14377] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #91 [Aug 18 10:34:12] DEBUG[14377] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Finding handler for 213049 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking channels create: Didn't match 213049 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:12] DEBUG[14368] res_ari.c: Checking channels externalMedia: Didn't match 213049 [Aug 18 10:34:12] DEBUG[14368] res_ari.c: No explicit handler found for 213049. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] VERBOSE[14372] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:12] VERBOSE[14372] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Initializing initreq for method INVITE - callid 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116934@178.62.121.41 SIP/2.0 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 3 [ 52]: From: ;tag=as126e0733 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 6 [ 60]: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:12 GMT [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:12] VERBOSE[14372] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #114 [Aug 18 10:34:12] DEBUG[14372] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[13953] chan_sip.c: Hangup call SIP/zvonobot-0000006c, SIP callid 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[13953] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:12] DEBUG[13953] res_rtp_asterisk.c: (0x7f0c30021550) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13953] res_rtp_asterisk.c: (0x7f0c30021550) DTLS srtp - stopped timeout timer' [Aug 18 10:34:12] DEBUG[13953] channel.c: Channel 0x7f0c3010c400 'SIP/zvonobot-0000006c' destroying [Aug 18 10:34:12] DEBUG[14256] channel.c: Channel 0x7f0c84147a20 'SIP/zvonobot-0000008e' allocated [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:12] DEBUG[14256] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:12] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[14115] channel.c: Channel 0x7f0c980440c0 'Announcer/ARI-00000027;1' destroying [Aug 18 10:34:12] DEBUG[14099] bridge_channel.c: Setting 0x7f0c980404a0(Announcer/ARI-00000027;2) state from:0 to:1 [Aug 18 10:34:12] DEBUG[14099] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: pulling 0x7f0c980404a0(Announcer/ARI-00000027;2) [Aug 18 10:34:12] VERBOSE[14099] bridge_channel.c: Channel Announcer/ARI-00000027;2 left 'softmix' stasis-bridge <0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3> [Aug 18 10:34:12] DEBUG[14099] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c980404a0(Announcer/ARI-00000027;2) is leaving softmix technology [Aug 18 10:34:12] DEBUG[14099] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3'. Checking compatability for channels 'SIP/zvonobot-0000002f' and 'Recorder/ARI-00000023;2' [Aug 18 10:34:12] DEBUG[14099] bridge_native_rtp.c: Bridge '0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3' can not use native RTP bridge as channel 'SIP/zvonobot-0000002f' has features which prevent it [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:12] DEBUG[14099] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:12] VERBOSE[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: switching from softmix technology to simple_bridge [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology constructor [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) to dummy bridge temporarily [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c20086d10(Recorder/ARI-00000023;2) to dummy bridge temporarily [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is leaving softmix technology (dummy) [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is leaving softmix technology (dummy) [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling softmix technology stop [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) is joining simple_bridge technology [Aug 18 10:34:12] DEBUG[14099] channel.c: Channel Recorder/ARI-00000023;2 setting read format path: slin -> slin [Aug 18 10:34:12] DEBUG[14099] channel.c: Channel Recorder/ARI-00000023;2 setting write format path: slin -> slin [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0c20086d10(Recorder/ARI-00000023;2) is joining simple_bridge technology [Aug 18 10:34:12] DEBUG[14115] stasis.c: Destroying topic. name: cache:272/channel:1629282843.231, detail: [Aug 18 10:34:12] DEBUG[14115] stasis.c: Topic 'cache:272/channel:1629282843.231': 0x7f0c98045ae0 destroyed [Aug 18 10:34:12] DEBUG[14115] stasis.c: Destroying topic. name: channel:1629282843.231, detail: [Aug 18 10:34:12] DEBUG[14115] stasis.c: Topic 'channel:1629282843.231': 0x7f0c980450b0 destroyed [Aug 18 10:34:12] DEBUG[14099] channel.c: Channel Recorder/ARI-00000023;2 setting read format path: slin -> slin [Aug 18 10:34:12] DEBUG[14353] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] VERBOSE[14377] dial.c: Called zvonobot/79821116936 [Aug 18 10:34:12] DEBUG[14256] res_stasis.c: calls_0: Subscribing to 213108 [Aug 18 10:34:12] DEBUG[14256] stasis/app.c: Channel '213108' is 1 interested in calls_0 [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117066@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2738b490 Max-Forwards: 70 From: ;tag=as410f495a To: ;tag=as5cb94950 Contact: Call-ID: 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:12] DEBUG[14353] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14099] channel.c: Channel Recorder/ARI-00000023;2 setting write format path: slin -> slin [Aug 18 10:34:12] DEBUG[14256] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:12] DEBUG[14256] http.c: HTTP closing session. Top level [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.335, detail: [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling simple_bridge technology start [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: deferring softmix technology destructor [Aug 18 10:34:12] DEBUG[14347] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3803fd80' [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:12] DEBUG[14099] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: queueing action type:13 sub:1000 [Aug 18 10:34:12] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP audio difference is 656, ms is 102 [Aug 18 10:34:12] DEBUG[13695] channel.c: SIP/zvonobot-0000002f: Dropping redundant connected line update "" <>. [Aug 18 10:34:12] DEBUG[20534] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:12] DEBUG[14113] bridge_softmix.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: stopping mixing thread [Aug 18 10:34:12] DEBUG[20534] bridge_softmix.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: Waiting for mixing thread to die. [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.335': 0x7f0c3013e980 created [Aug 18 10:34:12] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:388/channel:1629282852.335, detail: [Aug 18 10:34:12] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 800, ms is 70 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:388/channel:1629282852.335': 0x7f0c3007f570 created [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE add candidate: 10.131.0.10:15158, 2130706431 [Aug 18 10:34:12] VERBOSE[12933] dial.c: SIP/zvonobot-0000000a answered [Aug 18 10:34:12] DEBUG[13798] channel.c: Recorder/ARI-00000023;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:12] DEBUG[14379] chan_sip.c: Outgoing Call for 79821116932 [Aug 18 10:34:12] DEBUG[14379] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:12] DEBUG[14353] rtp_engine.c: RTP instance '0x7f0c7804d100' is setup and ready to go [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) ICE stopped [Aug 18 10:34:12] DEBUG[14353] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:12] DEBUG[14353] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213071': is 0 interested in calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis/app.c: channel '213071' unsubscribed from calls_0 [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: cache:270/channel:213071, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'cache:270/channel:213071': 0x7f0c3010eb40 destroyed [Aug 18 10:34:12] DEBUG[20620] stasis.c: Destroying topic. name: channel:213071, detail: [Aug 18 10:34:12] DEBUG[20620] stasis.c: Topic 'channel:213071': 0x7f0c3010e130 destroyed [Aug 18 10:34:12] VERBOSE[12933] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000000a [Aug 18 10:34:12] DEBUG[12933] stasis/app.c: Channel '212974' is 2 interested in calls_0 [Aug 18 10:34:12] VERBOSE[14372] dial.c: Called zvonobot/79821116934 [Aug 18 10:34:12] VERBOSE[12933] res_rtp_asterisk.c: 0x7f0c7800dea0 -- Strict RTP switching to RTP target address 178.62.121.41:11140 as source [Aug 18 10:34:12] DEBUG[12933] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:12] DEBUG[12933] channel.c: Channel SIP/zvonobot-0000000a setting read format path: ulaw -> alaw [Aug 18 10:34:12] DEBUG[12933] channel.c: Channel SIP/zvonobot-0000000a setting write format path: alaw -> ulaw [Aug 18 10:34:12] DEBUG[14353] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:12] DEBUG[14353] res_rtp_asterisk.c: (0x7f0c7804d100) RTCP setup on RTP instance [Aug 18 10:34:12] VERBOSE[14353] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:12] DEBUG[14353] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:12] WARNING[14133] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000028;1 [Aug 18 10:34:12] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP audio difference is 640, ms is 100 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Session timer started: 31 - 51b2b16964d5fe5b5593bebe09b5f7da@159.65.48.104:5060 1768000ms [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:12] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14353] chan_sip.c: SIP call-id changed from '2b8ca4bb22e45d7255b22df85bd105fb@127.0.1.1:5060' to '5129740f51f9292d29e823f263748e28@159.65.48.104:5060' [Aug 18 10:34:12] DEBUG[14099] channel.c: Channel 0x7f0c980b5300 'Announcer/ARI-00000027;2' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) RTP allocated port 14508 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (4) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (4) INVITE - 5 [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:12] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6383ms with no response [Aug 18 10:34:12] WARNING[20585] chan_sip.c: Hanging up call 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:12] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:388/channel:1629282852.335, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:388/channel:1629282852.335': 0x7f0c3007f570 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.335, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.335': 0x7f0c3013e980 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006b', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213067', '')] [Aug 18 10:34:12] DEBUG[14353] stasis.c: Creating topic. name: channel:213119, detail: [Aug 18 10:34:12] DEBUG[14353] stasis.c: Topic 'channel:213119': 0x7f0c78049400 created [Aug 18 10:34:12] DEBUG[14353] stasis.c: Creating topic. name: cache:389/channel:213119, detail: [Aug 18 10:34:12] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14353] stasis.c: Topic 'cache:389/channel:213119': 0x7f0c780240d0 created [Aug 18 10:34:12] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[14379] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:12] DEBUG[14379] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:12] VERBOSE[14379] chan_sip.c: Audio is at 14160 [Aug 18 10:34:12] DEBUG[14090] channel.c: Channel 0x7f0ca4040e00 'SIP/zvonobot-00000073' hanging up. Refs: 2 [Aug 18 10:34:12] DEBUG[14381] http.c: HTTP opening session. Top level [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.337, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.337': 0x7f0c3002e830 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:390/channel:1629282852.337, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:390/channel:1629282852.337': 0x7f0c30011950 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:390/channel:1629282852.337, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:390/channel:1629282852.337': 0x7f0c30011950 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.337, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.337': 0x7f0c3002e830 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001a', '', 'AppDial2', '(Outgoing Line)', 24, 0, 'BUSY', 3, '', '212988', '')] [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:12] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:12] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.338, detail: [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE creating session 0.0.0.0:14508 (14508) [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE create [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE add system candidates [Aug 18 10:34:12] DEBUG[14383] http.c: HTTP opening session. Top level [Aug 18 10:34:12] VERBOSE[14379] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:12] DEBUG[14383] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:12] DEBUG[14383] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14383] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[14383] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14383] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.338': 0x7f0c3002e830 created [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: cache:391/channel:1629282852.338, detail: [Aug 18 10:34:12] DEBUG[14381] http.c: HTTP Request URI is /ari/channels/213071 [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:391/channel:1629282852.338': 0x7f0c3007f570 created [Aug 18 10:34:12] DEBUG[14347] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:12] DEBUG[14347] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE add candidate: 159.65.48.104:14508, 2130706431 [Aug 18 10:34:12] DEBUG[14347] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:12] DEBUG[14347] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Finding handler for bridges [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:12] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE add candidate: 10.131.0.10:14508, 2130706431 [Aug 18 10:34:12] DEBUG[14381] http.c: match request [ari/channels/213071] with handler [httpstatus] len 10 [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:12] DEBUG[14381] http.c: match request [ari/channels/213071] with handler [phoneprov] len 9 [Aug 18 10:34:12] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: cache:391/channel:1629282852.338, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'cache:391/channel:1629282852.338': 0x7f0c3007f570 destroyed [Aug 18 10:34:12] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.338, detail: [Aug 18 10:34:12] DEBUG[20545] stasis.c: Topic 'channel:1629282852.338': 0x7f0c3002e830 destroyed [Aug 18 10:34:12] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000055', '', 'AppDial2', '(Outgoing Line)', 11, 0, 'NO ANSWER', 3, '', '213046', '')] [Aug 18 10:34:12] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:12] DEBUG[14381] http.c: match request [ari/channels/213071] with handler [ari] len 3 [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:12] DEBUG[14381] http.c: Match made with [ari] [Aug 18 10:34:12] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282852.339, detail: [Aug 18 10:34:12] VERBOSE[14379] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: Allocating new SIP dialog for 00cbda0541ae67c6648bf9a5276348e9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:13] DEBUG[14359] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c940cb950' [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282852.339': 0x7f0c3002e830 created [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:392/channel:1629282852.339, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:392/channel:1629282852.339': 0x7f0c30011950 created [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Finding handler for channels/213071 [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Finding handler for channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Finding handler for 213071 [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking channels create: Didn't match 213071 [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:13] DEBUG[14381] res_ari.c: Checking channels externalMedia: Didn't match 213071 [Aug 18 10:34:13] DEBUG[14381] res_ari.c: No explicit handler found for 213071. Using wildcard channelId. [Aug 18 10:34:12] DEBUG[14383] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) RTP allocated port 12644 [Aug 18 10:34:12] DEBUG[14347] rtp_engine.c: RTP instance '0x7f0c3803fd80' is setup and ready to go [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39c211e6;received=159.65.48.104 From: ;tag=as7cfb6ac0 To: ;tag=as15dd6f89 Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 752725804 752725804 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12658 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39c211e6;received=159.65.48.104 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7cfb6ac0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as15dd6f89 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 752725804 752725804 IN IP4 178.62.121.41 [Aug 18 10:34:13] VERBOSE[14379] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[14383] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:392/channel:1629282852.339, detail: [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE creating session 0.0.0.0:12644 (12644) [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:392/channel:1629282852.339': 0x7f0c30011950 destroyed [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE create [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE add system candidates [Aug 18 10:34:13] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE stopped [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12658 RTP/AVP 0 8 101 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282852.339, detail: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:13] DEBUG[14383] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282852.339': 0x7f0c3002e830 destroyed [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:13] DEBUG[14359] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:13] DEBUG[14359] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE add candidate: 159.65.48.104:12644, 2130706431 [Aug 18 10:34:13] DEBUG[14359] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:13] DEBUG[14359] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE add candidate: 10.131.0.10:12644, 2130706431 [Aug 18 10:34:13] DEBUG[14359] rtp_engine.c: RTP instance '0x7f0c940cb950' is setup and ready to go [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) ICE stopped [Aug 18 10:34:13] DEBUG[14359] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:13] DEBUG[14359] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:13] DEBUG[14359] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:13] DEBUG[14359] res_rtp_asterisk.c: (0x7f0c940cb950) RTCP setup on RTP instance [Aug 18 10:34:13] VERBOSE[14359] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:13] DEBUG[14359] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] DEBUG[14359] chan_sip.c: SIP call-id changed from '00cbda0541ae67c6648bf9a5276348e9@127.0.1.1:5060' to '007f90413610c97471d9f37255f670d0@159.65.48.104:5060' [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: = Looking for Call ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 (Checking To) --From tag as7cfb6ac0 --To-tag as15dd6f89 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Initializing initreq for method INVITE - callid 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116932@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 3 [ 52]: From: ;tag=as4a9f4c08 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 6 [ 60]: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:12 GMT [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14379] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #44 [Aug 18 10:34:13] DEBUG[14379] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000058', '', 'AppDial2', '(Outgoing Line)', 11, 0, 'NO ANSWER', 3, '', '213049', '')] [Aug 18 10:34:13] DEBUG[14236] channel.c: Channel 0x7f0c3800da10 'SIP/zvonobot-0000008f' allocated [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14236] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[14236] res_stasis.c: calls_0: Subscribing to 213107 [Aug 18 10:34:13] DEBUG[13171] res_rtp_asterisk.c: (0x7f0c38023850) RTP 0x7f0c380281e0 -- Received packet from 178.62.121.41:12658, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[14236] stasis/app.c: Channel '213107' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Outgoing Call for 79821116933 [Aug 18 10:34:13] DEBUG[14347] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:13] DEBUG[14236] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14236] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[14347] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:13] DEBUG[14347] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:13] DEBUG[14347] res_rtp_asterisk.c: (0x7f0c3803fd80) RTCP setup on RTP instance [Aug 18 10:34:13] VERBOSE[14347] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:13] DEBUG[14347] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] DEBUG[14347] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:13] DEBUG[14359] stasis.c: Creating topic. name: channel:213123, detail: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Stopping retransmission on '15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:13] DEBUG[14383] stasis.c: Creating topic. name: bridge:21515bb0-91f2-4ad5-852f-8721c870cad7, detail: [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006f - start 1629282845.787368 answer 0.000000 end 1629282852.306926 dur 6.519 bill 1629282852.306 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282853.341, detail: [Aug 18 10:34:13] VERBOSE[14379] dial.c: Called zvonobot/79821116932 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:13] DEBUG[14347] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:13] VERBOSE[14384] chan_sip.c: Audio is at 10398 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] VERBOSE[14384] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Got SDP version 752725804 and unique parts [root 752725804 IN IP4 178.62.121.41] [Aug 18 10:34:13] VERBOSE[14384] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 752725804 752725804 IN IP4 178.62.121.41... OK. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:13] VERBOSE[14384] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:13] WARNING[14273] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-0000002b;1 [Aug 18 10:34:13] DEBUG[14359] stasis.c: Topic 'channel:213123': 0x7f0c94029560 created [Aug 18 10:34:13] DEBUG[14359] stasis.c: Creating topic. name: cache:393/channel:213123, detail: [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.341': 0x7f0c3002e830 created [Aug 18 10:34:13] DEBUG[14347] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Initializing initreq for method INVITE - callid 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:394/channel:1629282853.341, detail: [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116933@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 3 [ 52]: From: ;tag=as52ec131c [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:394/channel:1629282853.341': 0x7f0c3007f570 created [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:394/channel:1629282853.341, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:394/channel:1629282853.341': 0x7f0c3007f570 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282853.341, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.341': 0x7f0c3002e830 destroyed [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000067', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213068', '')] [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 6 [ 60]: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14384] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14383] stasis.c: Topic 'bridge:21515bb0-91f2-4ad5-852f-8721c870cad7': 0x2c6fa90 created [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #37 [Aug 18 10:34:13] DEBUG[14359] stasis.c: Topic 'cache:393/channel:213123': 0x7f0c940cbea0 created [Aug 18 10:34:13] DEBUG[14384] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000068 - start 1629282845.586642 answer 0.000000 end 1629282852.356751 dur 6.770 bill 1629282852.356 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000006e - start 1629282845.687843 answer 0.000000 end 1629282852.371076 dur 6.683 bill 1629282852.371 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[14259] channel.c: Channel 0x7f0c80074680 'SIP/zvonobot-00000091' allocated [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14259] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14347] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] VERBOSE[14384] dial.c: Called zvonobot/79821116933 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:13] DEBUG[14259] res_stasis.c: calls_0: Subscribing to 213111 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[14383] stasis.c: Creating topic. name: cache:395/bridge:21515bb0-91f2-4ad5-852f-8721c870cad7, detail: [Aug 18 10:34:13] DEBUG[14259] stasis/app.c: Channel '213111' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14383] stasis.c: Topic 'cache:395/bridge:21515bb0-91f2-4ad5-852f-8721c870cad7': 0x2c35f70 created [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Outgoing Call for 79821116929 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[14259] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14259] http.c: HTTP closing session. Top level [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] VERBOSE[14386] chan_sip.c: Audio is at 19502 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000071 - start 1629282845.926460 answer 0.000000 end 1629282852.486090 dur 6.559 bill 1629282852.486 dispo NO ANSWER [Aug 18 10:34:13] VERBOSE[14386] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:13] VERBOSE[14386] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14347] chan_sip.c: SIP call-id changed from '696f5d62547db27838918a632a95bd69@127.0.1.1:5060' to '6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060' [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14383] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' can not use native RTP bridge as two channels are required [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000070 - start 1629282845.918355 answer 0.000000 end 1629282852.558584 dur 6.640 bill 1629282852.558 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38023850) ICE set role failed; no ice instance [Aug 18 10:34:13] VERBOSE[14386] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000072 - start 1629282845.977917 answer 0.000000 end 1629282852.677687 dur 6.699 bill 1629282852.677 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[14383] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Initializing initreq for method INVITE - callid 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116929@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38023850) RTCP setting address on RTP instance [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 3 [ 52]: From: ;tag=as48744f9a [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 6 [ 60]: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14347] stasis.c: Creating topic. name: channel:213120, detail: [Aug 18 10:34:13] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c380281e0 -- Strict RTP learning after remote address set to: 178.62.121.41:12658 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12658 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0048db8) from 0x7f0c147e2330 to 0x7f0c38023a28 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14383] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb000ac48) from 0x7f0c147e2330 to 0x7f0c38023a28 [Aug 18 10:34:13] VERBOSE[14386] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Aug 18 10:34:13] DEBUG[14386] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282853.343, detail: [Aug 18 10:34:13] VERBOSE[14386] dial.c: Called zvonobot/79821116929 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb011dbf8) from 0x7f0c147e2330 to 0x7f0c38023a28 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38023850) RTCP ignoring duplicate property [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[14383] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.343': 0x7f0c3002e830 created [Aug 18 10:34:13] DEBUG[14347] stasis.c: Topic 'channel:213120': 0x7f0c3808fbb0 created [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:396/channel:1629282853.343, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:396/channel:1629282853.343': 0x7f0c3007f570 created [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:396/channel:1629282853.343, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:396/channel:1629282853.343': 0x7f0c3007f570 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282853.343, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.343': 0x7f0c3002e830 destroyed [Aug 18 10:34:13] DEBUG[14383] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[14383] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: calling simple_bridge technology constructor [Aug 18 10:34:13] DEBUG[14347] stasis.c: Creating topic. name: cache:397/channel:213120, detail: [Aug 18 10:34:13] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000029 setting read format path: alaw -> alaw [Aug 18 10:34:13] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006c', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213071', '')] [Aug 18 10:34:13] DEBUG[14383] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: calling simple_bridge technology start [Aug 18 10:34:13] DEBUG[14347] stasis.c: Topic 'cache:397/channel:213120': 0x7f0c38033170 created [Aug 18 10:34:13] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000029 setting write format path: alaw -> alaw [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38023850) DTLS - ast_rtp_activate rtp=0x7f0c380281e0 - setup and perform DTLS' [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c380281e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c380281e0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:13] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP got report of 100 bytes from 178.62.121.41:10225 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Strict routing enforced for session 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117034@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e1d8b2 Max-Forwards: 70 From: ;tag=as7cfb6ac0 To: ;tag=as15dd6f89 Contact: Call-ID: 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (1) INVITE - 5 [Aug 18 10:34:13] VERBOSE[13171] dial.c: SIP/zvonobot-00000029 answered [Aug 18 10:34:13] VERBOSE[13171] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000029 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:13] DEBUG[13171] stasis/app.c: Channel '213006' is 2 interested in calls_0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (4) INVITE - 5 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:13] VERBOSE[13171] res_rtp_asterisk.c: 0x7f0c380281e0 -- Strict RTP switching to RTP target address 178.62.121.41:12658 as source [Aug 18 10:34:13] DEBUG[13171] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:13] DEBUG[13171] channel.c: Channel SIP/zvonobot-00000029 setting read format path: ulaw -> alaw [Aug 18 10:34:13] DEBUG[13171] channel.c: Channel SIP/zvonobot-00000029 setting write format path: alaw -> ulaw [Aug 18 10:34:13] DEBUG[14389] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14389] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:13] DEBUG[14389] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14389] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14389] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[14390] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14389] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[14251] channel.c: Channel 0x7f0c78053fb0 'SIP/zvonobot-00000090' allocated [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14251] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[14383] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2485aced650f4f671041baca16773141@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:13] DEBUG[14390] http.c: HTTP Request URI is /ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/addChannel?channel=212974 [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30021550) DTLS stop [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[14389] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[14389] stasis.c: Creating topic. name: bridge:fa4500bd-5aa0-4ef0-bd71-2993d47cd759, detail: [Aug 18 10:34:13] DEBUG[14390] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14389] stasis.c: Topic 'bridge:fa4500bd-5aa0-4ef0-bd71-2993d47cd759': 0x7f0c1c147ca0 created [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30021550) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000073 - start 1629282846.297307 answer 0.000000 end 1629282852.920223 dur 6.622 bill 1629282852.920 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[14390] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[14383] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14389] stasis.c: Creating topic. name: cache:398/bridge:fa4500bd-5aa0-4ef0-bd71-2993d47cd759, detail: [Aug 18 10:34:13] DEBUG[14389] stasis.c: Topic 'cache:398/bridge:fa4500bd-5aa0-4ef0-bd71-2993d47cd759': 0x7f0c1c0671d0 created [Aug 18 10:34:13] DEBUG[14390] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/addChannel] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30021550) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[14390] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30021550) ICE RTP transport deallocating [Aug 18 10:34:13] DEBUG[14389] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' can not use native RTP bridge as two channels are required [Aug 18 10:34:13] DEBUG[14389] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c30021550' [Aug 18 10:34:13] DEBUG[14389] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Finding handler for bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/addChannel [Aug 18 10:34:13] DEBUG[14389] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:13] DEBUG[14389] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[14389] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling simple_bridge technology constructor [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[14389] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling simple_bridge technology start [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #116 (5) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #116)) [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Finding handler for 21515bb0-91f2-4ad5-852f-8721c870cad7 [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116949@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263 Max-Forwards: 70 From: ;tag=as7d784780 To: Contact: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1222004691 1222004691 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18068 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14389] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14389] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:13] DEBUG[14390] res_ari.c: No explicit handler found for 21515bb0-91f2-4ad5-852f-8721c870cad7. Using wildcard bridgeId. [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Finding handler for addChannel [Aug 18 10:34:13] DEBUG[14390] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:13] DEBUG[14390] stasis/control.c: 212974: Sending channel add_to_bridge command [Aug 18 10:34:13] DEBUG[14391] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14391] http.c: HTTP Request URI is /ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/addChannel?channel=213006 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14391] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14391] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (4) INVITE - 5 [Aug 18 10:34:13] DEBUG[14391] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/addChannel] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[12933] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000000a [Aug 18 10:34:13] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14260] channel.c: Channel 0x7f0c8c11cb60 'SIP/zvonobot-00000092' allocated [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14260] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14391] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Finding handler for bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/addChannel [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[12933] stasis/control.c: 212974: Adding to bridge 21515bb0-91f2-4ad5-852f-8721c870cad7 [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #6 (4) INVITE - 5 [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #6)) [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 768, ms is 68 [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Finding handler for fa4500bd-5aa0-4ef0-bd71-2993d47cd759 [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:13] DEBUG[14391] res_ari.c: No explicit handler found for fa4500bd-5aa0-4ef0-bd71-2993d47cd759. Using wildcard bridgeId. [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Finding handler for addChannel [Aug 18 10:34:13] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[12933] stasis/app.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14266] channel.c: Channel 0x7f0c900a0ad0 'SIP/zvonobot-00000093' allocated [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14266] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[14392] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: 0x7f0c84082e50(SIP/zvonobot-0000000a) is joining [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (4) INVITE - 5 [Aug 18 10:34:13] DEBUG[14391] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14391] stasis/control.c: 213006: Sending channel add_to_bridge command [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:13] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Session timer started: 124 - 15a504d457192cb8464cfb1246d479ee@159.65.48.104:5060 1768000ms [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (2) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14251] res_stasis.c: calls_0: Subscribing to 213109 [Aug 18 10:34:13] DEBUG[14251] stasis/app.c: Channel '213109' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14251] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[13171] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000029 [Aug 18 10:34:13] DEBUG[13171] stasis/control.c: 213006: Adding to bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759 [Aug 18 10:34:13] DEBUG[13171] stasis/app.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14251] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14394] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40072e10(SIP/zvonobot-00000029) is joining [Aug 18 10:34:13] DEBUG[14260] res_stasis.c: calls_0: Subscribing to 213112 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9;received=159.65.48.104 From: ;tag=as6ac21020 To: ;tag=as01e0c440 Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="346d51ec" Content-Length: 0 <-------------> [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9;received=159.65.48.104 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ac21020 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as01e0c440 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="346d51ec" [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 (Checking To) --From tag as6ac21020 --To-tag as01e0c440 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (2) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14392] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: pushing 0x7f0c84082e50(SIP/zvonobot-0000000a) [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (4) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6203ms with no response [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Hanging up call 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: Allocating new SIP dialog for 1d512b10283bbe29789f2df96325e499@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:13] DEBUG[14355] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8006ad00' [Aug 18 10:34:13] DEBUG[14260] stasis/app.c: Channel '213112' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14260] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14260] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14266] res_stasis.c: calls_0: Subscribing to 213113 [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Outgoing Call for 79821116928 [Aug 18 10:34:13] VERBOSE[14392] bridge_channel.c: Channel SIP/zvonobot-0000000a joined 'simple_bridge' stasis-bridge <21515bb0-91f2-4ad5-852f-8721c870cad7> [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14117] channel.c: Channel 0x7f0cac095830 'SIP/zvonobot-00000074' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Outgoing Call for 79821116931 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) RTP allocated port 15044 [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] DEBUG[14266] stasis/app.c: Channel '213113' is 1 interested in calls_0 [Aug 18 10:34:13] VERBOSE[14395] chan_sip.c: Audio is at 15988 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000074 - start 1629282846.927982 answer 0.000000 end 1629282853.435129 dur 6.507 bill 1629282853.435 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (2) INVITE - 5 [Aug 18 10:34:13] VERBOSE[14395] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] DEBUG[14266] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14266] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Outgoing Call for 79821116927 [Aug 18 10:34:13] VERBOSE[14395] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[14394] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: pushing 0x7f0c40072e10(SIP/zvonobot-00000029) [Aug 18 10:34:13] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] VERBOSE[14395] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] VERBOSE[14393] chan_sip.c: Audio is at 11680 [Aug 18 10:34:13] VERBOSE[14394] bridge_channel.c: Channel SIP/zvonobot-00000029 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:13] VERBOSE[14393] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[14392] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' can not use native RTP bridge as two channels are required [Aug 18 10:34:13] DEBUG[14392] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[14392] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (2) INVITE - 5 [Aug 18 10:34:13] DEBUG[14392] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Initializing initreq for method INVITE - callid 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116928@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] VERBOSE[14396] chan_sip.c: Audio is at 12990 [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE creating session 0.0.0.0:15044 (15044) [Aug 18 10:34:13] DEBUG[14392] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[14392] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7 is already using the new technology. [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 3 [ 52]: From: ;tag=as03ee25b2 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:34:13] VERBOSE[14396] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[14393] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 6 [ 60]: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] VERBOSE[14396] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14392] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: 0x7f0c84082e50(SIP/zvonobot-0000000a) is joining simple_bridge technology [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] VERBOSE[14396] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14395] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] VERBOSE[14393] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #61 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[14395] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[14395] dial.c: Called zvonobot/79821116928 [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE create [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[12933] stasis/app.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' is 2 interested in calls_0 [Aug 18 10:34:13] DEBUG[14390] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[14390] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14397] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Initializing initreq for method INVITE - callid 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116927@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 3 [ 52]: From: ;tag=as2c4993ac [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14392] res_rtp_asterisk.c: (0x7f0c7800c760) RTP changing ssrc from 2109335315 to 1877870525 due to a source change [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 6 [ 60]: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 From: ;tag=as22c76af6 To: ;tag=as509aa30f Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 389657146 389657146 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10944 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as509aa30f [Aug 18 10:34:13] DEBUG[14397] http.c: HTTP Request URI is /ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/record?name=212974_FudEJPETWNVEuxovCtssfLyerOQQfoOM&format=wav [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Initializing initreq for method INVITE - callid 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116931@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 3 [ 52]: From: ;tag=as3bdaa5eb [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 6 [ 60]: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14393] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Aug 18 10:34:13] DEBUG[14393] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 389657146 389657146 IN IP4 178.62.121.41 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10944 RTP/AVP 0 8 101 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag as509aa30f [Aug 18 10:34:13] VERBOSE[14396] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE add system candidates [Aug 18 10:34:13] VERBOSE[14393] dial.c: Called zvonobot/79821116931 [Aug 18 10:34:13] DEBUG[14397] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/record] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #60 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Stopping retransmission on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Got SDP version 389657146 and unique parts [root 389657146 IN IP4 178.62.121.41] [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 389657146 389657146 IN IP4 178.62.121.41... OK. [Aug 18 10:34:13] DEBUG[14394] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' can not use native RTP bridge as two channels are required [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[14396] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:13] DEBUG[14397] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/record] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[14263] channel.c: Channel 0x7f0c940d16b0 'SIP/zvonobot-00000094' allocated [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[14263] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:13] VERBOSE[14396] dial.c: Called zvonobot/79821116927 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:13] DEBUG[14263] res_stasis.c: calls_0: Subscribing to 213110 [Aug 18 10:34:13] DEBUG[14263] stasis/app.c: Channel '213110' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Outgoing Call for 79821116930 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] VERBOSE[14400] chan_sip.c: Audio is at 15638 [Aug 18 10:34:13] VERBOSE[14400] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] VERBOSE[14400] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] VERBOSE[14400] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Initializing initreq for method INVITE - callid 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116930@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 3 [ 52]: From: ;tag=as14ba6e32 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 6 [ 60]: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14400] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Aug 18 10:34:13] DEBUG[14400] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[14400] dial.c: Called zvonobot/79821116930 [Aug 18 10:34:13] DEBUG[14263] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[14263] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14394] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[14397] http.c: match request [ari/bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/record] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[14394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:13] WARNING[14156] app.c: No audio available on Recorder/ARI-00000029;1?? [Aug 18 10:34:13] VERBOSE[14156] app.c: User hung up [Aug 18 10:34:13] DEBUG[14156] res_stasis_recording.c: 1629282843.237: Recording complete [Aug 18 10:34:13] DEBUG[14156] channel.c: Channel 0x7f0c78090610 'Recorder/ARI-00000029;1' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[14355] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:13] DEBUG[14355] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE add candidate: 159.65.48.104:15044, 2130706431 [Aug 18 10:34:13] DEBUG[14355] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:13] DEBUG[14355] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE add candidate: 10.131.0.10:15044, 2130706431 [Aug 18 10:34:13] DEBUG[14355] rtp_engine.c: RTP instance '0x7f0c8006ad00' is setup and ready to go [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE stopped [Aug 18 10:34:13] DEBUG[14355] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:13] DEBUG[14355] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:13] DEBUG[14355] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[14397] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14394] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:13] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14394] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14355] res_rtp_asterisk.c: (0x7f0c8006ad00) RTCP setup on RTP instance [Aug 18 10:34:13] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14394] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759 is already using the new technology. [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Finding handler for bridges/21515bb0-91f2-4ad5-852f-8721c870cad7/record [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:13] DEBUG[14394] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40072e10(SIP/zvonobot-00000029) is joining simple_bridge technology [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:13] WARNING[13741] app.c: No audio available on Recorder/ARI-00000020;1?? [Aug 18 10:34:13] VERBOSE[14355] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:13] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 16 instead [Aug 18 10:34:13] VERBOSE[13741] app.c: User hung up [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[13741] res_stasis_recording.c: 1629282839.181: Recording complete [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[13741] channel.c: Channel 0x7f0c2c08ce90 'Recorder/ARI-00000020;1' hanging up. Refs: 2 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14394] res_rtp_asterisk.c: (0x7f0c38023850) RTP changing ssrc from 646267711 to 1745782646 due to a source change [Aug 18 10:34:13] DEBUG[13171] stasis/app.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' is 2 interested in calls_0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:13] DEBUG[14391] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:13] DEBUG[14391] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:13] DEBUG[14355] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14403] http.c: HTTP opening session. Top level [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[14403] http.c: HTTP Request URI is /ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/record?name=213006_JPrwZVkIlHRZZBNMHYoLeMMJKYBGrECr&format=wav [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca804b700) ICE set role failed; no ice instance [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:13] DEBUG[14355] chan_sip.c: SIP call-id changed from '1d512b10283bbe29789f2df96325e499@127.0.1.1:5060' to '7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060' [Aug 18 10:34:13] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Finding handler for 21515bb0-91f2-4ad5-852f-8721c870cad7 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca804b700) RTCP setting address on RTP instance [Aug 18 10:34:13] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0ca8050e80 -- Strict RTP learning after remote address set to: 178.62.121.41:10944 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10944 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0089b08) from 0x7f0c147e2330 to 0x7f0ca804b8d8 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00c42d8) from 0x7f0c147e2330 to 0x7f0ca804b8d8 [Aug 18 10:34:13] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb01045e8) from 0x7f0c147e2330 to 0x7f0ca804b8d8 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca804b700) RTCP ignoring duplicate property [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:13] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000047 setting read format path: alaw -> alaw [Aug 18 10:34:13] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000047 setting write format path: alaw -> alaw [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca804b700) DTLS - ast_rtp_activate rtp=0x7f0ca8050e80 - setup and perform DTLS' [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8050e80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8050e80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Strict routing enforced for session 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:13] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:13] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117004@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0473a0ca Max-Forwards: 70 From: ;tag=as22c76af6 To: ;tag=as509aa30f Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14403] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/record] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:13] VERBOSE[13542] dial.c: SIP/zvonobot-00000047 answered [Aug 18 10:34:13] DEBUG[14355] stasis.c: Creating topic. name: channel:213121, detail: [Aug 18 10:34:13] DEBUG[14355] stasis.c: Topic 'channel:213121': 0x7f0c80038640 created [Aug 18 10:34:13] DEBUG[14355] stasis.c: Creating topic. name: cache:399/channel:213121, detail: [Aug 18 10:34:13] DEBUG[14355] stasis.c: Topic 'cache:399/channel:213121': 0x7f0c800314b0 created [Aug 18 10:34:13] DEBUG[14403] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/record] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[14403] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/record] with handler [ari] len 3 [Aug 18 10:34:13] VERBOSE[13542] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000047 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: No explicit handler found for 21515bb0-91f2-4ad5-852f-8721c870cad7. Using wildcard bridgeId. [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Finding handler for record [Aug 18 10:34:13] DEBUG[14403] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[13542] stasis/app.c: Channel '213036' is 2 interested in calls_0 [Aug 18 10:34:13] DEBUG[13563] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP 0x7f0ca4101500 -- Received packet from 178.62.121.41:15868, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[13563] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP 0x7f0ca4101500 -- Received packet from 178.62.121.41:15868, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Finding handler for bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/record [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:13] DEBUG[14404] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14404] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:13] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (2) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Finding handler for fa4500bd-5aa0-4ef0-bd71-2993d47cd759 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14159] channel.c: Channel 0x7f0c1c127e20 'Announcer/ARI-00000028;1' destroying [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:13] DEBUG[14239] channel.c: Channel 0x7f0c7405a610 'SIP/zvonobot-00000095' allocated [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 564726e17074235c1af6801638e43e42@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6375ms with no response [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:13] DEBUG[14404] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14403] res_ari.c: No explicit handler found for fa4500bd-5aa0-4ef0-bd71-2993d47cd759. Using wildcard bridgeId. [Aug 18 10:34:13] DEBUG[14133] bridge_channel.c: Setting 0x7f0c1c136810(Announcer/ARI-00000028;2) state from:0 to:1 [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Hanging up call 564726e17074235c1af6801638e43e42@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 564726e17074235c1af6801638e43e42@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14239] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:13] DEBUG[14133] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pulling 0x7f0c1c136810(Announcer/ARI-00000028;2) [Aug 18 10:34:13] VERBOSE[14133] bridge_channel.c: Channel Announcer/ARI-00000028;2 left 'softmix' stasis-bridge [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Finding handler for record [Aug 18 10:34:13] DEBUG[14404] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[14159] stasis.c: Destroying topic. name: cache:275/channel:1629282843.234, detail: [Aug 18 10:34:13] DEBUG[14159] stasis.c: Topic 'cache:275/channel:1629282843.234': 0x7f0c1c136e80 destroyed [Aug 18 10:34:13] DEBUG[14133] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c1c136810(Announcer/ARI-00000028;2) is leaving softmix technology [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14159] stasis.c: Destroying topic. name: channel:1629282843.234, detail: [Aug 18 10:34:13] DEBUG[14397] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:13] DEBUG[14159] stasis.c: Topic 'channel:1629282843.234': 0x7f0c1c136420 destroyed [Aug 18 10:34:13] DEBUG[14404] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:13] DEBUG[14403] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:13] DEBUG[14119] channel.c: Channel 0x7f0c9c08d3f0 'SIP/zvonobot-00000075' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000075 - start 1629282847.065482 answer 0.000000 end 1629282853.727541 dur 6.662 bill 1629282853.727 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6366ms with no response [Aug 18 10:34:13] DEBUG[14403] stasis.c: Creating topic. name: channel:1629282853.346, detail: [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:13] DEBUG[14133] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52'. Checking compatability for channels 'SIP/zvonobot-00000030' and 'Recorder/ARI-0000001c;2' [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:13] DEBUG[14133] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as channel 'SIP/zvonobot-00000030' has features which prevent it [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14133] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] VERBOSE[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: switching from softmix technology to simple_bridge [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology constructor [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0ca0073e00(SIP/zvonobot-00000030) to dummy bridge temporarily [Aug 18 10:34:13] DEBUG[14404] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0c78074930(Recorder/ARI-0000001c;2) to dummy bridge temporarily [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is leaving softmix technology (dummy) [Aug 18 10:34:13] DEBUG[13659] chan_sip.c: Hangup call SIP/zvonobot-00000057, SIP callid 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14091] channel.c: Channel 0x7f0c180f8f60 'Announcer/ARI-00000032;1' allocated [Aug 18 10:34:13] DEBUG[14091] stasis.c: Creating topic. name: channel:1629282853.347, detail: [Aug 18 10:34:13] DEBUG[14091] stasis.c: Topic 'channel:1629282853.347': 0x7f0c180c7a30 created [Aug 18 10:34:13] DEBUG[14091] stasis.c: Creating topic. name: cache:400/channel:1629282853.347, detail: [Aug 18 10:34:13] DEBUG[14091] stasis.c: Topic 'cache:400/channel:1629282853.347': 0x7f0c180a9610 created [Aug 18 10:34:13] DEBUG[14273] bridge_channel.c: Setting 0x7f0c900b05e0(Announcer/ARI-0000002b;2) state from:0 to:1 [Aug 18 10:34:13] DEBUG[14273] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: pulling 0x7f0c900b05e0(Announcer/ARI-0000002b;2) [Aug 18 10:34:13] VERBOSE[14273] bridge_channel.c: Channel Announcer/ARI-0000002b;2 left 'softmix' stasis-bridge <95aa254a-8cb0-4e7f-94b3-e5d21f2bb060> [Aug 18 10:34:13] DEBUG[14273] bridge_channel.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c900b05e0(Announcer/ARI-0000002b;2) is leaving softmix technology [Aug 18 10:34:13] WARNING[20585] chan_sip.c: Hanging up call 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is leaving softmix technology (dummy) [Aug 18 10:34:13] DEBUG[14290] channel.c: Channel 0x7f0c900818f0 'Announcer/ARI-0000002b;1' destroying [Aug 18 10:34:13] DEBUG[13659] res_rtp_asterisk.c: (0x7f0c180c9c20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[13659] res_rtp_asterisk.c: (0x7f0c180c9c20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[13659] channel.c: Channel 0x7f0c180d2b30 'SIP/zvonobot-00000057' destroying [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology stop [Aug 18 10:34:13] DEBUG[14278] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is joining simple_bridge technology [Aug 18 10:34:13] DEBUG[14133] channel.c: Channel Recorder/ARI-0000001c;2 setting read format path: slin -> slin [Aug 18 10:34:13] DEBUG[14133] channel.c: Channel Recorder/ARI-0000001c;2 setting write format path: slin -> slin [Aug 18 10:34:13] DEBUG[20535] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Aug 18 10:34:13] DEBUG[13669] chan_sip.c: Hangup call SIP/zvonobot-00000059, SIP callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[13981] chan_sip.c: Hangup call SIP/zvonobot-0000006d, SIP callid 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14278] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14298] channel.c: Channel 0x7f0c40071ab0 'Recorder/ARI-00000033;1' allocated [Aug 18 10:34:13] DEBUG[14298] stasis.c: Creating topic. name: channel:1629282853.348, detail: [Aug 18 10:34:13] DEBUG[13669] res_rtp_asterisk.c: (0x7f0c340ab160) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[14397] stasis.c: Creating topic. name: channel:1629282853.345, detail: [Aug 18 10:34:13] DEBUG[13981] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:13] DEBUG[13669] res_rtp_asterisk.c: (0x7f0c340ab160) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[13669] channel.c: Channel 0x7f0c340bc830 'SIP/zvonobot-00000059' destroying [Aug 18 10:34:13] DEBUG[13563] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP 0x7f0ca4101500 -- Received packet from 178.62.121.41:15868, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[13563] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP 0x7f0ca4101500 -- Received packet from 178.62.121.41:15868, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[13563] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP 0x7f0ca4101500 -- Received packet from 178.62.121.41:15868, dropping due to strict RTP protection. [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50291 [Aug 18 10:34:13] DEBUG[20616] app_queue.c: Device 'Announcer/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:13] DEBUG[13981] res_rtp_asterisk.c: (0x7f0c38082a80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[14298] stasis.c: Topic 'channel:1629282853.348': 0x7f0c400a6b40 created [Aug 18 10:34:13] DEBUG[14298] stasis.c: Creating topic. name: cache:401/channel:1629282853.348, detail: [Aug 18 10:34:13] DEBUG[14298] stasis.c: Topic 'cache:401/channel:1629282853.348': 0x7f0c4005b170 created [Aug 18 10:34:13] DEBUG[13981] res_rtp_asterisk.c: (0x7f0c38082a80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[13981] channel.c: Channel 0x7f0c38099400 'SIP/zvonobot-0000006d' destroying [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213045': is 0 interested in calls_0 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213045' unsubscribed from calls_0 [Aug 18 10:34:13] DEBUG[14403] stasis.c: Topic 'channel:1629282853.346': 0x7f0c40067ae0 created [Aug 18 10:34:13] DEBUG[14122] channel.c: Channel 0x7f0cb4080db0 'SIP/zvonobot-00000076' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is joining simple_bridge technology [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282853.349, detail: [Aug 18 10:34:13] DEBUG[14403] stasis.c: Creating topic. name: cache:402/channel:1629282853.346, detail: [Aug 18 10:34:13] DEBUG[14405] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14403] stasis.c: Topic 'cache:402/channel:1629282853.346': 0x7f0c400a3d80 created [Aug 18 10:34:13] DEBUG[14290] stasis.c: Destroying topic. name: cache:303/channel:1629282846.257, detail: [Aug 18 10:34:13] DEBUG[14405] http.c: HTTP Request URI is /ari/channels/213045 [Aug 18 10:34:13] DEBUG[13659] stasis.c: Destroying topic. name: channel:213045, detail: [Aug 18 10:34:13] DEBUG[13659] stasis.c: Topic 'channel:213045': 0x7f0c180c4730 destroyed [Aug 18 10:34:13] DEBUG[14290] stasis.c: Topic 'cache:303/channel:1629282846.257': 0x7f0c9004ec10 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.349': 0x7f0c3005adf0 created [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[14273] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060'. Checking compatability for channels 'SIP/zvonobot-0000002b' and 'Recorder/ARI-0000001a;2' [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:403/channel:1629282853.349, detail: [Aug 18 10:34:13] DEBUG[20620] stasis.c: Destroying topic. name: cache:198/channel:213045, detail: [Aug 18 10:34:13] DEBUG[14273] bridge_native_rtp.c: Bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' can not use native RTP bridge as channel 'SIP/zvonobot-0000002b' has features which prevent it [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:13] DEBUG[14405] http.c: match request [ari/channels/213045] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14405] http.c: match request [ari/channels/213045] with handler [phoneprov] len 9 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:403/channel:1629282853.349': 0x7f0c3007f570 created [Aug 18 10:34:13] DEBUG[14133] channel.c: Channel Recorder/ARI-0000001c;2 setting read format path: slin -> slin [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[20620] stasis.c: Topic 'cache:198/channel:213045': 0x7f0c1808f6e0 destroyed [Aug 18 10:34:13] DEBUG[14273] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[14405] http.c: match request [ari/channels/213045] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 688, ms is 63 [Aug 18 10:34:13] DEBUG[14290] stasis.c: Destroying topic. name: channel:1629282846.257, detail: [Aug 18 10:34:13] VERBOSE[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: switching from softmix technology to simple_bridge [Aug 18 10:34:13] DEBUG[14290] stasis.c: Topic 'channel:1629282846.257': 0x7f0c90010b00 destroyed [Aug 18 10:34:13] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology constructor [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c7803b960(SIP/zvonobot-0000002b) to dummy bridge temporarily [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[14405] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:403/channel:1629282853.349, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:403/channel:1629282853.349': 0x7f0c3007f570 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282853.349, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.349': 0x7f0c3005adf0 destroyed [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000057', '', 'AppDial2', '(Outgoing Line)', 11, 0, 'NO ANSWER', 3, '', '213045', '')] [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: moving 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) to dummy bridge temporarily [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is leaving softmix technology (dummy) [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is leaving softmix technology (dummy) [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology stop [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c7803b960(SIP/zvonobot-0000002b) is joining simple_bridge technology [Aug 18 10:34:13] DEBUG[14397] stasis.c: Topic 'channel:1629282853.345': 0x7f0c3c119170 created [Aug 18 10:34:13] DEBUG[14133] channel.c: Channel Recorder/ARI-0000001c;2 setting write format path: slin -> slin [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology start [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: deferring softmix technology destructor [Aug 18 10:34:13] DEBUG[14133] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: queueing action type:13 sub:1000 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282853.350, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.350': 0x7f0c3005e3e0 created [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:405/channel:1629282853.350, detail: [Aug 18 10:34:13] DEBUG[14273] channel.c: Channel Recorder/ARI-0000001a;2 setting read format path: slin -> slin [Aug 18 10:34:13] DEBUG[14273] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: 0x7f0c8004d4e0(Recorder/ARI-0000001a;2) is joining simple_bridge technology [Aug 18 10:34:13] DEBUG[14273] channel.c: Channel Recorder/ARI-0000001a;2 setting read format path: slin -> slin [Aug 18 10:34:13] DEBUG[14273] channel.c: Channel Recorder/ARI-0000001a;2 setting write format path: slin -> slin [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology start [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: deferring softmix technology destructor [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213072': is 0 interested in calls_0 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213072' unsubscribed from calls_0 [Aug 18 10:34:13] DEBUG[20620] stasis.c: Destroying topic. name: cache:271/channel:213072, detail: [Aug 18 10:34:13] DEBUG[20620] stasis.c: Topic 'cache:271/channel:213072': 0x7f0c3809bbd0 destroyed [Aug 18 10:34:13] DEBUG[20620] stasis.c: Destroying topic. name: channel:213072, detail: [Aug 18 10:34:13] DEBUG[20620] stasis.c: Topic 'channel:213072': 0x7f0c3809b150 destroyed [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213050': is 0 interested in calls_0 [Aug 18 10:34:13] DEBUG[20620] stasis/app.c: channel '213050' unsubscribed from calls_0 [Aug 18 10:34:13] DEBUG[20620] stasis.c: Destroying topic. name: cache:201/channel:213050, detail: [Aug 18 10:34:13] DEBUG[20620] stasis.c: Topic 'cache:201/channel:213050': 0x7f0c340bf030 destroyed [Aug 18 10:34:13] DEBUG[14273] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: queueing action type:13 sub:1000 [Aug 18 10:34:13] DEBUG[14407] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Finding handler for bridges [Aug 18 10:34:13] DEBUG[14406] http.c: HTTP opening session. Top level [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:405/channel:1629282853.350': 0x7f0c300e53f0 created [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Finding handler for channels/213045 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Session timer started: 73 - 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 1768000ms [Aug 18 10:34:13] DEBUG[14406] http.c: HTTP Request URI is /ari/channels/213072 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Finding handler for channels [Aug 18 10:34:13] DEBUG[14406] http.c: match request [ari/channels/213072] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14407] http.c: HTTP Request URI is /ari/channels/213050 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:13] DEBUG[14407] http.c: match request [ari/channels/213050] with handler [httpstatus] len 10 [Aug 18 10:34:13] DEBUG[14407] http.c: match request [ari/channels/213050] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 848, ms is 73 [Aug 18 10:34:13] DEBUG[14239] res_stasis.c: calls_0: Subscribing to 213105 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:13] DEBUG[14239] stasis/app.c: Channel '213105' is 1 interested in calls_0 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:13] DEBUG[14406] http.c: match request [ari/channels/213072] with handler [phoneprov] len 9 [Aug 18 10:34:13] DEBUG[14239] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:405/channel:1629282853.350, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:405/channel:1629282853.350': 0x7f0c300e53f0 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282853.350, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.350': 0x7f0c3005e3e0 destroyed [Aug 18 10:34:13] DEBUG[14239] http.c: HTTP closing session. Top level [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000059', '', 'AppDial2', '(Outgoing Line)', 11, 0, 'NO ANSWER', 3, '', '213050', '')] [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Outgoing Call for 79821116935 [Aug 18 10:34:13] DEBUG[14406] http.c: match request [ari/channels/213072] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[14407] http.c: match request [ari/channels/213050] with handler [ari] len 3 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Finding handler for 213045 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking channels create: Didn't match 213045 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14405] res_ari.c: Checking channels externalMedia: Didn't match 213045 [Aug 18 10:34:13] DEBUG[14405] res_ari.c: No explicit handler found for 213045. Using wildcard channelId. [Aug 18 10:34:13] DEBUG[14406] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:13] DEBUG[14397] stasis.c: Creating topic. name: cache:404/channel:1629282853.345, detail: [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Finding handler for channels/213072 [Aug 18 10:34:13] DEBUG[13626] channel.c: Recorder/ARI-0000001a;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Finding handler for channels [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:13] DEBUG[14407] http.c: Match made with [ari] [Aug 18 10:34:13] DEBUG[14288] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: stopping mixing thread [Aug 18 10:34:13] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:13] DEBUG[20620] stasis.c: Destroying topic. name: channel:213050, detail: [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Finding handler for channels/213050 [Aug 18 10:34:13] DEBUG[20620] stasis.c: Topic 'channel:213050': 0x7f0c340be5b0 destroyed [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Finding handler for 213072 [Aug 18 10:34:13] DEBUG[14273] channel.c: Channel 0x7f0c90025910 'Announcer/ARI-0000002b;2' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[13678] channel.c: Recorder/ARI-0000001c;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282853.351, detail: [Aug 18 10:34:13] DEBUG[13556] channel.c: SIP/zvonobot-0000002b: Dropping redundant connected line update "" <>. [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking channels create: Didn't match 213072 [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:13] DEBUG[14154] bridge_softmix.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: stopping mixing thread [Aug 18 10:34:13] DEBUG[20534] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Finding handler for channels [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:13] DEBUG[14406] res_ari.c: Checking channels externalMedia: Didn't match 213072 [Aug 18 10:34:13] DEBUG[20534] bridge_softmix.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: Waiting for mixing thread to die. [Aug 18 10:34:13] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP audio difference is 1472, ms is 204 [Aug 18 10:34:13] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:13] DEBUG[20534] bridge_softmix.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: Waiting for mixing thread to die. [Aug 18 10:34:13] DEBUG[14397] stasis.c: Topic 'cache:404/channel:1629282853.345': 0x7f0c3c004d90 created [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:13] DEBUG[13619] channel.c: SIP/zvonobot-00000030: Dropping redundant connected line update "" <>. [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.351': 0x7f0c300fba90 created [Aug 18 10:34:13] DEBUG[14406] res_ari.c: No explicit handler found for 213072. Using wildcard channelId. [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:13] DEBUG[20545] stasis.c: Creating topic. name: cache:406/channel:1629282853.351, detail: [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:406/channel:1629282853.351': 0x7f0c3007f570 created [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:13] VERBOSE[14408] chan_sip.c: Audio is at 14706 [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:13] DEBUG[14404] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 From: ;tag=as04f0121c To: ;tag=as31dfa2de Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as04f0121c [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31dfa2de [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 (Checking To) --From tag as04f0121c --To-tag as31dfa2de [Aug 18 10:34:13] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: cache:406/channel:1629282853.351, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'cache:406/channel:1629282853.351': 0x7f0c3007f570 destroyed [Aug 18 10:34:13] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282853.351, detail: [Aug 18 10:34:13] DEBUG[20545] stasis.c: Topic 'channel:1629282853.351': 0x7f0c300fba90 destroyed [Aug 18 10:34:13] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:04', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006d', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213072', '')] [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:13] DEBUG[14133] channel.c: Channel 0x7f0c1c13de00 'Announcer/ARI-00000028;2' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:13] DEBUG[14404] stasis.c: Creating topic. name: bridge:61075423-3ee2-4d60-8382-ee99e654a5be, detail: [Aug 18 10:34:13] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] VERBOSE[14408] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Stopping retransmission on '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:13] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Finding handler for 213050 [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking channels create: Didn't match 213050 [Aug 18 10:34:13] VERBOSE[14408] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae Max-Forwards: 70 From: ;tag=as04f0121c To: ;tag=as31dfa2de Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:13] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000076 - start 1629282847.281264 answer 0.000000 end 1629282853.764009 dur 6.482 bill 1629282853.764 dispo NO ANSWER [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:13] VERBOSE[13167] dial.c: SIP/zvonobot-00000028 is busy [Aug 18 10:34:13] DEBUG[13167] channel.c: Channel 0x7f0c30038fd0 'SIP/zvonobot-00000028' hanging up. Refs: 2 [Aug 18 10:34:13] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:13] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:13] VERBOSE[14408] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Initializing initreq for method INVITE - callid 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116935@178.62.121.41 SIP/2.0 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 3 [ 52]: From: ;tag=as3a399b13 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 6 [ 60]: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:13 GMT [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:13] VERBOSE[14408] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #107 [Aug 18 10:34:13] DEBUG[14408] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:13] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000028 - start 1629282830.005443 answer 0.000000 end 1629282853.973409 dur 23.967 bill 1629282853.973 dispo BUSY [Aug 18 10:34:13] DEBUG[14404] stasis.c: Topic 'bridge:61075423-3ee2-4d60-8382-ee99e654a5be': 0x7f0c74097e90 created [Aug 18 10:34:13] DEBUG[14404] stasis.c: Creating topic. name: cache:407/bridge:61075423-3ee2-4d60-8382-ee99e654a5be, detail: [Aug 18 10:34:13] DEBUG[14404] stasis.c: Topic 'cache:407/bridge:61075423-3ee2-4d60-8382-ee99e654a5be': 0x7f0c74071020 created [Aug 18 10:34:13] DEBUG[14404] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' can not use native RTP bridge as two channels are required [Aug 18 10:34:13] DEBUG[14404] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:13] DEBUG[14404] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:13] DEBUG[14404] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:13] DEBUG[14404] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:13] DEBUG[14404] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: calling simple_bridge technology constructor [Aug 18 10:34:13] DEBUG[14404] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: calling simple_bridge technology start [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (1) INVITE - 5 [Aug 18 10:34:13] DEBUG[14200] stasis.c: Creating topic. name: channel:1629282853.352, detail: [Aug 18 10:34:13] DEBUG[14200] stasis.c: Topic 'channel:1629282853.352': 0x7f0ca00582c0 created [Aug 18 10:34:13] DEBUG[14200] stasis.c: Creating topic. name: cache:408/channel:1629282853.352, detail: [Aug 18 10:34:13] DEBUG[14200] stasis.c: Topic 'cache:408/channel:1629282853.352': 0x7f0ca00eb620 created [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:13] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:13] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:13] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:13] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (3) INVITE - 5 [Aug 18 10:34:13] DEBUG[14407] res_ari.c: Checking channels externalMedia: Didn't match 213050 [Aug 18 10:34:14] DEBUG[13552] res_rtp_asterisk.c: (0x7f0c980a1440) RTP 0x7f0c980a2f60 -- Received packet from 178.62.121.41:12380, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14407] res_ari.c: No explicit handler found for 213050. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14136] channel.c: Channel 0x7f0c9c0ab760 'Announcer/ARI-0000002f;2' allocated [Aug 18 10:34:14] DEBUG[14136] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:14] DEBUG[14136] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000002f;1' [Aug 18 10:34:14] DEBUG[14416] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) is joining [Aug 18 10:34:14] DEBUG[14414] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14414] http.c: HTTP Request URI is /ari/channels/213124?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116916&callerId=74950493843 [Aug 18 10:34:14] DEBUG[14414] http.c: match request [ari/channels/213124] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14414] http.c: match request [ari/channels/213124] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14416] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pushing 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) [Aug 18 10:34:14] DEBUG[14414] http.c: match request [ari/channels/213124] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14414] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14414] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Finding handler for channels/213124 [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14302] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14302] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 656, ms is 61 [Aug 18 10:34:14] DEBUG[14272] res_rtp_asterisk.c: (0x7f0c98083570) RTP audio difference is 784, ms is 69 [Aug 18 10:34:14] DEBUG[14404] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116944@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120 Max-Forwards: 70 From: ;tag=as22d5765f To: Contact: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1895674049 1895674049 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (2) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (2) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[14416] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:14] VERBOSE[14416] bridge_channel.c: Channel Announcer/ARI-0000002f;2 joined 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14416] bridge.c: Chose bridge technology softmix [Aug 18 10:34:14] VERBOSE[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: switching from simple_bridge technology to softmix [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology constructor [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c18091350(SIP/zvonobot-00000013) to dummy bridge temporarily [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c940389d0(Recorder/ARI-00000024;2) to dummy bridge temporarily [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is leaving simple_bridge technology (dummy) [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology stop [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) is joining softmix technology [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Announcer/ARI-0000002f;2: [Aug 18 10:34:14] DEBUG[14416] channel.c: Channel Announcer/ARI-0000002f;2 setting write format path: slin -> slin [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Announcer/ARI-0000002f;2: [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Announcer/ARI-0000002f;2: Not in SFU mode [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining softmix technology [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: SIP/zvonobot-00000013: [Aug 18 10:34:14] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: SIP/zvonobot-00000013: [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Finding handler for 213124 [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking channels create: Didn't match 213124 [Aug 18 10:34:14] DEBUG[14419] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[13768] res_rtp_asterisk.c: (0x7f0ca804bf40) RTP audio difference is 720, ms is 65 [Aug 18 10:34:14] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14418] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: SIP/zvonobot-00000013: Not in SFU mode [Aug 18 10:34:14] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14414] res_ari.c: Checking channels externalMedia: Didn't match 213124 [Aug 18 10:34:14] DEBUG[14414] res_ari.c: No explicit handler found for 213124. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is joining softmix technology [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Recorder/ARI-00000024;2: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (2) INVITE - 5 [Aug 18 10:34:14] DEBUG[14416] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: slin -> slin [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14416] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Recorder/ARI-00000024;2: [Aug 18 10:34:14] DEBUG[14416] bridge_softmix.c: Recorder/ARI-00000024;2: Not in SFU mode [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology start [Aug 18 10:34:14] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology destructor [Aug 18 10:34:14] DEBUG[14404] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:14] DEBUG[14423] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14423] http.c: HTTP Request URI is /ari/channels/213126?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116914&callerId=74950493843 [Aug 18 10:34:14] DEBUG[14431] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14431] http.c: HTTP Request URI is /ari/channels/213127?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116913&callerId=74950493843 [Aug 18 10:34:14] VERBOSE[14408] dial.c: Called zvonobot/79821116935 [Aug 18 10:34:14] DEBUG[14431] http.c: match request [ari/channels/213127] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14431] http.c: match request [ari/channels/213127] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14431] http.c: match request [ari/channels/213127] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14431] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14431] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Finding handler for channels/213127 [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Finding handler for 213127 [Aug 18 10:34:14] DEBUG[14418] http.c: HTTP Request URI is /ari/channels/213125?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116915&callerId=74950493843 [Aug 18 10:34:14] DEBUG[14419] http.c: HTTP Request URI is /ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/addChannel?channel=213036 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking channels create: Didn't match 213127 [Aug 18 10:34:14] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14431] res_ari.c: Checking channels externalMedia: Didn't match 213127 [Aug 18 10:34:14] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14431] res_ari.c: No explicit handler found for 213127. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14418] http.c: match request [ari/channels/213125] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14211] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14136] res_stasis_playback.c: 1629282847.269: Sending play(sound:silence/2) command [Aug 18 10:34:14] DEBUG[14136] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:14] DEBUG[14211] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14136] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14418] http.c: match request [ari/channels/213125] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14418] http.c: match request [ari/channels/213125] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14418] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14418] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14419] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[14420] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: starting mixing thread [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116942@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ac3289e Max-Forwards: 70 From: ;tag=as001c84c2 To: Contact: Call-ID: 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2025029083 2025029083 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15826 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[14423] http.c: match request [ari/channels/213126] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14419] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Finding handler for channels/213125 [Aug 18 10:34:14] DEBUG[14419] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/addChannel] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6490ms with no response [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Hanging up call 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Finding handler for 213125 [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking channels create: Didn't match 213125 [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14418] res_ari.c: Checking channels externalMedia: Didn't match 213125 [Aug 18 10:34:14] DEBUG[14418] res_ari.c: No explicit handler found for 213125. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14433] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14419] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[13704] audiohook.c: Audiohook 0x7f0c8805b620 has stale audio in its factories. Flushing them both [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:14] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000077 - start 1629282847.512714 answer 0.000000 end 1629282854.156936 dur 6.644 bill 1629282854.156 dispo NO ANSWER [Aug 18 10:34:14] DEBUG[14141] channel.c: Channel 0x7f0cb0160ed0 'SIP/zvonobot-00000077' hanging up. Refs: 2 [Aug 18 10:34:14] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Destroying SIP dialog 54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14434] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14423] http.c: match request [ari/channels/213126] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14433] http.c: HTTP Request URI is /ari/channels/213128?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116912&callerId=74950493843 [Aug 18 10:34:14] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP ooh, format changed from none to ulaw [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:14] DEBUG[14435] http.c: HTTP opening session. Top level [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '54be89257b1b9e8f3286232023c58de8@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:14] DEBUG[14433] http.c: match request [ari/channels/213128] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14423] http.c: match request [ari/channels/213126] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14433] http.c: match request [ari/channels/213128] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14435] http.c: HTTP Request URI is /ari/channels/213130?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116910&callerId=74950493843 [Aug 18 10:34:14] DEBUG[14434] http.c: HTTP Request URI is /ari/channels/213129?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116911&callerId=74950493843 [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14423] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: Allocating new SIP dialog for 322c0694628c2a717e628c22322fa905@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14414] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8007aae0' [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) RTP allocated port 14784 [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE creating session 0.0.0.0:14784 (14784) [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE create [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14433] http.c: match request [ari/channels/213128] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14423] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14433] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180c9c20) DTLS stop [Aug 18 10:34:14] DEBUG[14435] http.c: match request [ari/channels/213130] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14433] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Finding handler for channels/213128 [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Finding handler for 213128 [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking channels create: Didn't match 213128 [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14433] res_ari.c: Checking channels externalMedia: Didn't match 213128 [Aug 18 10:34:14] DEBUG[14433] res_ari.c: No explicit handler found for 213128. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180c9c20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14331] app.c: One waitfor failed, trying another [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Finding handler for channels/213126 [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Finding handler for 213126 [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking channels create: Didn't match 213126 [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14423] res_ari.c: Checking channels externalMedia: Didn't match 213126 [Aug 18 10:34:14] DEBUG[14423] res_ari.c: No explicit handler found for 213126. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14434] http.c: match request [ari/channels/213129] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14434] http.c: match request [ari/channels/213129] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14434] http.c: match request [ari/channels/213129] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14434] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14434] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Finding handler for channels/213129 [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Finding handler for 213129 [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking channels create: Didn't match 213129 [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14434] res_ari.c: Checking channels externalMedia: Didn't match 213129 [Aug 18 10:34:14] DEBUG[14434] res_ari.c: No explicit handler found for 213129. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180c9c20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180c9c20) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c180c9c20' [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ab160) DTLS stop [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ab160) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ab160) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340ab160) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c340ab160' [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Destroying SIP dialog 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38082a80) DTLS stop [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38082a80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38082a80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38082a80) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c38082a80' [Aug 18 10:34:14] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14435] http.c: match request [ari/channels/213130] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Finding handler for bridges/61075423-3ee2-4d60-8382-ee99e654a5be/addChannel [Aug 18 10:34:14] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 From: ;tag=as57df1d1c To: ;tag=as31c5c295 Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7961f4b1;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57df1d1c [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31c5c295 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 (Checking To) --From tag as57df1d1c --To-tag as31c5c295 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 31862e904180bb0f5f1cd1dd172b82a6@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (2) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116943@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456 Max-Forwards: 70 From: ;tag=as307f6396 To: Contact: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1191644094 1191644094 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (1) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 From: ;tag=as79336d5f To: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 From: ;tag=as63ca65c0 To: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f;received=159.65.48.104 From: ;tag=as123045f1 To: ;tag=as2329ece7 Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as123045f1 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2329ece7 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 (Checking To) --From tag as123045f1 --To-tag as2329ece7 [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14436] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14435] http.c: match request [ari/channels/213130] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Stopping retransmission on '5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117062@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2088996f Max-Forwards: 70 From: ;tag=as123045f1 To: ;tag=as2329ece7 Contact: Call-ID: 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14436] http.c: HTTP Request URI is /ari/channels/213131?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116909&callerId=74950493843 [Aug 18 10:34:14] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] VERBOSE[12965] dial.c: SIP/zvonobot-00000011 is busy [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14435] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14435] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Finding handler for channels/213130 [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Finding handler for 213130 [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking channels create: Didn't match 213130 [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14435] res_ari.c: Checking channels externalMedia: Didn't match 213130 [Aug 18 10:34:14] DEBUG[14435] res_ari.c: No explicit handler found for 213130. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000011 - start 1629282824.193367 answer 0.000000 end 1629282854.243260 dur 30.049 bill 1629282854.243 dispo BUSY [Aug 18 10:34:14] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE add system candidates [Aug 18 10:34:14] DEBUG[14414] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14414] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE add candidate: 159.65.48.104:14784, 2130706431 [Aug 18 10:34:14] DEBUG[14414] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14414] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE add candidate: 10.131.0.10:14784, 2130706431 [Aug 18 10:34:14] DEBUG[12965] channel.c: Channel 0x7f0cb401fdb0 'SIP/zvonobot-00000011' hanging up. Refs: 2 [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Finding handler for bridges [Aug 18 10:34:14] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14436] http.c: match request [ari/channels/213131] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:14] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 720, ms is 65 [Aug 18 10:34:14] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 640, ms is 60 [Aug 18 10:34:14] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14436] http.c: match request [ari/channels/213131] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: Allocating new SIP dialog for 51fa180b48c62b2e22cb8c3c36277c15@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14433] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca0108a20' [Aug 18 10:34:14] DEBUG[14436] http.c: match request [ari/channels/213131] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14436] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14436] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Finding handler for channels/213131 [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Finding handler for 213131 [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking channels create: Didn't match 213131 [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14436] res_ari.c: Checking channels externalMedia: Didn't match 213131 [Aug 18 10:34:14] DEBUG[14436] res_ari.c: No explicit handler found for 213131. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:14] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:14] DEBUG[14432] channel.c: Channel Announcer/ARI-0000002f;1 setting write format path: gsm -> slin [Aug 18 10:34:14] DEBUG[14432] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:14] VERBOSE[14432] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14207] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[13996] channel.c: Channel 0x7f0c20015500 'Announcer/ARI-00000031;2' allocated [Aug 18 10:34:14] DEBUG[13996] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:14] DEBUG[13996] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000031;1' [Aug 18 10:34:14] DEBUG[14207] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14414] rtp_engine.c: RTP instance '0x7f0c8007aae0' is setup and ready to go [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757;received=159.65.48.104 From: ;tag=as02885f54 To: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as02885f54 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 (Checking To) --From tag as02885f54 --To-tag [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (2) INVITE - 5 [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) RTP allocated port 10372 [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE creating session 0.0.0.0:10372 (10372) [Aug 18 10:34:14] DEBUG[14443] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:14] DEBUG[14440] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[13994] chan_sip.c: Hangup call SIP/zvonobot-0000006a, SIP callid 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 640, ms is 60 [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: Allocating new SIP dialog for 09ecd4ec5376a8c5524285784b73dc91@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14431] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c09bf60' [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) RTP allocated port 11634 [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE creating session 0.0.0.0:11634 (11634) [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE create [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE add system candidates [Aug 18 10:34:14] DEBUG[14431] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14431] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE add candidate: 159.65.48.104:11634, 2130706431 [Aug 18 10:34:14] DEBUG[14431] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14431] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE add candidate: 10.131.0.10:11634, 2130706431 [Aug 18 10:34:14] DEBUG[14431] rtp_engine.c: RTP instance '0x7f0c9c09bf60' is setup and ready to go [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE stopped [Aug 18 10:34:14] DEBUG[14431] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14431] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14431] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE create [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE add system candidates [Aug 18 10:34:14] DEBUG[14433] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14433] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE add candidate: 159.65.48.104:10372, 2130706431 [Aug 18 10:34:14] DEBUG[14433] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14433] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE add candidate: 10.131.0.10:10372, 2130706431 [Aug 18 10:34:14] DEBUG[14433] rtp_engine.c: RTP instance '0x7f0ca0108a20' is setup and ready to go [Aug 18 10:34:14] DEBUG[14443] http.c: HTTP Request URI is /ari/channels/213132?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116908&callerId=74950493843 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[13994] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:14] DEBUG[13994] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13994] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13994] channel.c: Channel 0x7f0c2c0c8bd0 'SIP/zvonobot-0000006a' destroying [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14443] http.c: match request [ari/channels/213132] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14444] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c20073c60(Announcer/ARI-00000031;2) is joining [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE stopped [Aug 18 10:34:14] DEBUG[14414] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14414] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14414] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14414] res_rtp_asterisk.c: (0x7f0c8007aae0) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14414] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14414] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14414] chan_sip.c: SIP call-id changed from '322c0694628c2a717e628c22322fa905@127.0.1.1:5060' to '2749fa7d41ec862f1556002a63546011@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14414] stasis.c: Creating topic. name: channel:213124, detail: [Aug 18 10:34:14] DEBUG[14414] stasis.c: Topic 'channel:213124': 0x7f0c8009fc60 created [Aug 18 10:34:14] DEBUG[14414] stasis.c: Creating topic. name: cache:409/channel:213124, detail: [Aug 18 10:34:14] DEBUG[14414] stasis.c: Topic 'cache:409/channel:213124': 0x7f0c800a06e0 created [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Finding handler for 61075423-3ee2-4d60-8382-ee99e654a5be [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282854.354, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.354': 0x7f0c300fba90 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: cache:410/channel:1629282854.354, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:410/channel:1629282854.354': 0x7f0c300a4990 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: cache:410/channel:1629282854.354, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:410/channel:1629282854.354': 0x7f0c300a4990 destroyed [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282854.354, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.354': 0x7f0c300fba90 destroyed [Aug 18 10:34:14] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006a', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213070', '')] [Aug 18 10:34:14] DEBUG[14431] res_rtp_asterisk.c: (0x7f0c9c09bf60) RTCP setup on RTP instance [Aug 18 10:34:14] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14419] res_ari.c: No explicit handler found for 61075423-3ee2-4d60-8382-ee99e654a5be. Using wildcard bridgeId. [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Finding handler for addChannel [Aug 18 10:34:14] DEBUG[14419] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:14] DEBUG[14419] stasis/control.c: 213036: Sending channel add_to_bridge command [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6250ms with no response [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Hanging up call 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE stopped [Aug 18 10:34:14] DEBUG[14433] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14433] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14433] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14433] res_rtp_asterisk.c: (0x7f0ca0108a20) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14433] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14433] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14433] chan_sip.c: SIP call-id changed from '51fa180b48c62b2e22cb8c3c36277c15@127.0.1.1:5060' to '79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14433] stasis.c: Creating topic. name: channel:213128, detail: [Aug 18 10:34:14] DEBUG[14433] stasis.c: Topic 'channel:213128': 0x7f0ca00767f0 created [Aug 18 10:34:14] DEBUG[14433] stasis.c: Creating topic. name: cache:411/channel:213128, detail: [Aug 18 10:34:14] DEBUG[14433] stasis.c: Topic 'cache:411/channel:213128': 0x7f0ca00fd6c0 created [Aug 18 10:34:14] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel '213070': is 0 interested in calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel '213070' unsubscribed from calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: cache:266/channel:213070, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'cache:266/channel:213070': 0x7f0c2c0cb790 destroyed [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: channel:213070, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'channel:213070': 0x7f0c2c0cad10 destroyed [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007a - start 1629282848.002416 answer 0.000000 end 1629282854.345404 dur 6.342 bill 1629282854.345 dispo NO ANSWER [Aug 18 10:34:14] VERBOSE[14431] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14431] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14431] chan_sip.c: SIP call-id changed from '09ecd4ec5376a8c5524285784b73dc91@127.0.1.1:5060' to '517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14431] stasis.c: Creating topic. name: channel:213127, detail: [Aug 18 10:34:14] DEBUG[14431] stasis.c: Topic 'channel:213127': 0x7f0c9c0609e0 created [Aug 18 10:34:14] DEBUG[14431] stasis.c: Creating topic. name: cache:412/channel:213127, detail: [Aug 18 10:34:14] DEBUG[14431] stasis.c: Topic 'cache:412/channel:213127': 0x7f0c9c025bd0 created [Aug 18 10:34:14] DEBUG[14165] channel.c: Channel 0x7f0ca0104fa0 'SIP/zvonobot-0000007a' hanging up. Refs: 2 [Aug 18 10:34:14] DEBUG[14446] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14446] http.c: HTTP Request URI is /ari/channels/213070 [Aug 18 10:34:14] DEBUG[14443] http.c: match request [ari/channels/213132] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14440] http.c: HTTP Request URI is /ari/channels/213133?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116907&callerId=74950493843 [Aug 18 10:34:14] DEBUG[13317] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14443] http.c: match request [ari/channels/213132] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14443] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14440] http.c: match request [ari/channels/213133] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14440] http.c: match request [ari/channels/213133] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14440] http.c: match request [ari/channels/213133] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14440] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14440] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: Allocating new SIP dialog for 7cbb077133582d4004d256e43cca0e19@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Destroying SIP dialog 436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '436656fe1ce2eb1047e0f502538ac008@159.65.48.104:5060' Method: BYE [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS stop [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c94022610) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c94022610' [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0;received=159.65.48.104 From: ;tag=as080d6dff To: ;tag=as5181b3f0 Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78be2a15" Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as080d6dff [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5181b3f0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78be2a15" [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 (Checking To) --From tag as080d6dff --To-tag as5181b3f0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6269ms with no response [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Hanging up call 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14443] http.c: HTTP consuming request body [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: Allocating new SIP dialog for 57bbce7049d24fba21ad2fa61d6bf925@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14434] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca40fbc00' [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) RTP allocated port 17238 [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE creating session 0.0.0.0:17238 (17238) [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE create [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE add system candidates [Aug 18 10:34:14] DEBUG[14434] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14434] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE add candidate: 159.65.48.104:17238, 2130706431 [Aug 18 10:34:14] DEBUG[14434] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14434] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE add candidate: 10.131.0.10:17238, 2130706431 [Aug 18 10:34:14] DEBUG[14434] rtp_engine.c: RTP instance '0x7f0ca40fbc00' is setup and ready to go [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE stopped [Aug 18 10:34:14] DEBUG[14434] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14434] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14434] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14434] res_rtp_asterisk.c: (0x7f0ca40fbc00) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14434] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14434] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14434] chan_sip.c: SIP call-id changed from '57bbce7049d24fba21ad2fa61d6bf925@127.0.1.1:5060' to '11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14434] stasis.c: Creating topic. name: channel:213129, detail: [Aug 18 10:34:14] DEBUG[14434] stasis.c: Topic 'channel:213129': 0x7f0ca4123eb0 created [Aug 18 10:34:14] DEBUG[14434] stasis.c: Creating topic. name: cache:413/channel:213129, detail: [Aug 18 10:34:14] DEBUG[14434] stasis.c: Topic 'cache:413/channel:213129': 0x7f0ca4124930 created [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: Allocating new SIP dialog for 36142a542538dd9a612f248a1c343900@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14435] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0161a90' [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) RTP allocated port 15334 [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE creating session 0.0.0.0:15334 (15334) [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE create [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE add system candidates [Aug 18 10:34:14] DEBUG[14435] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14435] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE add candidate: 159.65.48.104:15334, 2130706431 [Aug 18 10:34:14] DEBUG[14435] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14435] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE add candidate: 10.131.0.10:15334, 2130706431 [Aug 18 10:34:14] DEBUG[14435] rtp_engine.c: RTP instance '0x7f0cb0161a90' is setup and ready to go [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE stopped [Aug 18 10:34:14] DEBUG[14435] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14435] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14435] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14435] res_rtp_asterisk.c: (0x7f0cb0161a90) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14435] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14435] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14435] chan_sip.c: SIP call-id changed from '36142a542538dd9a612f248a1c343900@127.0.1.1:5060' to '233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14435] stasis.c: Creating topic. name: channel:213130, detail: [Aug 18 10:34:14] DEBUG[14435] stasis.c: Topic 'channel:213130': 0x7f0cb01270c0 created [Aug 18 10:34:14] DEBUG[14435] stasis.c: Creating topic. name: cache:414/channel:213130, detail: [Aug 18 10:34:14] DEBUG[14435] stasis.c: Topic 'cache:414/channel:213130': 0x7f0cb0079d30 created [Aug 18 10:34:14] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Finding handler for channels/213133 [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 736, ms is 66 [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14446] http.c: match request [ari/channels/213070] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 720, ms is 65 [Aug 18 10:34:14] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 688, ms is 63 [Aug 18 10:34:14] DEBUG[14161] channel.c: Channel 0x7f0c9c09ffc0 'SIP/zvonobot-00000079' hanging up. Refs: 2 [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000079 - start 1629282847.794646 answer 0.000000 end 1629282854.396928 dur 6.602 bill 1629282854.396 dispo NO ANSWER [Aug 18 10:34:14] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 832, ms is 72 [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Finding handler for channels/213132 [Aug 18 10:34:14] DEBUG[14446] http.c: match request [ari/channels/213070] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP audio difference is 784, ms is 118 [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:14] DEBUG[14423] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca80ebb80' [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Finding handler for 213132 [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking channels create: Didn't match 213132 [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14443] res_ari.c: Checking channels externalMedia: Didn't match 213132 [Aug 18 10:34:14] DEBUG[14443] res_ari.c: No explicit handler found for 213132. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) RTP allocated port 12208 [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE creating session 0.0.0.0:12208 (12208) [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE create [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE add system candidates [Aug 18 10:34:14] DEBUG[14423] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14423] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE add candidate: 159.65.48.104:12208, 2130706431 [Aug 18 10:34:14] DEBUG[14423] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14423] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE add candidate: 10.131.0.10:12208, 2130706431 [Aug 18 10:34:14] DEBUG[14423] rtp_engine.c: RTP instance '0x7f0ca80ebb80' is setup and ready to go [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE stopped [Aug 18 10:34:14] DEBUG[14423] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14423] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14423] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14423] res_rtp_asterisk.c: (0x7f0ca80ebb80) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14423] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14423] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14423] chan_sip.c: SIP call-id changed from '7cbb077133582d4004d256e43cca0e19@127.0.1.1:5060' to '7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14423] stasis.c: Creating topic. name: channel:213126, detail: [Aug 18 10:34:14] DEBUG[14423] stasis.c: Topic 'channel:213126': 0x7f0ca81095c0 created [Aug 18 10:34:14] DEBUG[14423] stasis.c: Creating topic. name: cache:415/channel:213126, detail: [Aug 18 10:34:14] DEBUG[14423] stasis.c: Topic 'cache:415/channel:213126': 0x7f0ca81188b0 created [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Destroying SIP dialog 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '31084e6149d402b41e86a7dd14209045@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: Allocating new SIP dialog for 260247d95b03ab1f77fdede61d27af23@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14436] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac01e130' [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) RTP allocated port 10524 [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE creating session 0.0.0.0:10524 (10524) [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE create [Aug 18 10:34:14] DEBUG[14446] http.c: match request [ari/channels/213070] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add system candidates [Aug 18 10:34:14] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:14] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14446] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Finding handler for 213133 [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking channels create: Didn't match 213133 [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Finding handler for channels/213070 [Aug 18 10:34:14] DEBUG[14444] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pushing 0x7f0c20073c60(Announcer/ARI-00000031;2) [Aug 18 10:34:14] DEBUG[14440] res_ari.c: Checking channels externalMedia: Didn't match 213133 [Aug 18 10:34:14] DEBUG[13512] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS stop [Aug 18 10:34:14] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14440] res_ari.c: No explicit handler found for 213133. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[13542] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000047 [Aug 18 10:34:14] DEBUG[13542] stasis/control.c: 213036: Adding to bridge 61075423-3ee2-4d60-8382-ee99e654a5be [Aug 18 10:34:14] DEBUG[13542] stasis/app.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' is 1 interested in calls_0 [Aug 18 10:34:14] DEBUG[14209] channel.c: Soft-Hanging (0x20) up channel 'SIP/zvonobot-00000000' [Aug 18 10:34:14] DEBUG[14436] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14447] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0c180c67b0(SIP/zvonobot-00000047) is joining [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14436] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 159.65.48.104:10524, 2130706431 [Aug 18 10:34:14] DEBUG[14436] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14436] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 10.131.0.10:10524, 2130706431 [Aug 18 10:34:14] DEBUG[14436] rtp_engine.c: RTP instance '0x7f0cac01e130' is setup and ready to go [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) ICE stopped [Aug 18 10:34:14] DEBUG[14436] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14436] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14436] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14436] res_rtp_asterisk.c: (0x7f0cac01e130) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14436] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14436] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14436] chan_sip.c: SIP call-id changed from '260247d95b03ab1f77fdede61d27af23@127.0.1.1:5060' to '47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14436] stasis.c: Creating topic. name: channel:213131, detail: [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[13676] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c0a9e10' [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14209] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:14] DEBUG[14209] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14444] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:14] VERBOSE[14444] bridge_channel.c: Channel Announcer/ARI-00000031;2 joined 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:34:14] DEBUG[14447] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: pushing 0x7f0c180c67b0(SIP/zvonobot-00000047) [Aug 18 10:34:14] DEBUG[14448] http.c: HTTP opening session. Top level [Aug 18 10:34:14] VERBOSE[14447] bridge_channel.c: Channel SIP/zvonobot-00000047 joined 'simple_bridge' stasis-bridge <61075423-3ee2-4d60-8382-ee99e654a5be> [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Finding handler for 213070 [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking channels create: Didn't match 213070 [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14446] res_ari.c: Checking channels externalMedia: Didn't match 213070 [Aug 18 10:34:14] DEBUG[14446] res_ari.c: No explicit handler found for 213070. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14448] http.c: HTTP Request URI is /ari/channels/1629282827.33 [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14444] bridge.c: Chose bridge technology softmix [Aug 18 10:34:14] VERBOSE[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: switching from simple_bridge technology to softmix [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology constructor [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0ca803dbf0(SIP/zvonobot-0000003b) to dummy bridge temporarily [Aug 18 10:34:14] DEBUG[14447] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' can not use native RTP bridge as two channels are required [Aug 18 10:34:14] DEBUG[14447] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:14] DEBUG[14447] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14436] stasis.c: Topic 'channel:213131': 0x7f0cac0740b0 created [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: moving 0x7f0c94055bb0(Recorder/ARI-0000001e;2) to dummy bridge temporarily [Aug 18 10:34:14] DEBUG[14447] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14448] http.c: match request [ari/channels/1629282827.33] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14447] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:14] DEBUG[14448] http.c: match request [ari/channels/1629282827.33] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[14447] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be is already using the new technology. [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: Allocating new SIP dialog for 27b89e867da32d886d1883b57c8eb1ec@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14418] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c88038c40' [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) RTP allocated port 17970 [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE creating session 0.0.0.0:17970 (17970) [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE create [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE add system candidates [Aug 18 10:34:14] DEBUG[14418] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14418] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE add candidate: 159.65.48.104:17970, 2130706431 [Aug 18 10:34:14] DEBUG[14418] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14418] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE add candidate: 10.131.0.10:17970, 2130706431 [Aug 18 10:34:14] DEBUG[14418] rtp_engine.c: RTP instance '0x7f0c88038c40' is setup and ready to go [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) ICE stopped [Aug 18 10:34:14] DEBUG[14418] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14418] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is leaving simple_bridge technology (dummy) [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:14] DEBUG[14447] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0c180c67b0(SIP/zvonobot-00000047) is joining simple_bridge technology [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology stop [Aug 18 10:34:14] DEBUG[14448] http.c: match request [ari/channels/1629282827.33] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14436] stasis.c: Creating topic. name: cache:416/channel:213131, detail: [Aug 18 10:34:14] DEBUG[14418] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14418] res_rtp_asterisk.c: (0x7f0c88038c40) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14418] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14219] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c20073c60(Announcer/ARI-00000031;2) is joining softmix technology [Aug 18 10:34:14] DEBUG[13542] stasis/app.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' is 2 interested in calls_0 [Aug 18 10:34:14] DEBUG[14219] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Setting 0x7f0cac00a6f0(SIP/zvonobot-00000000) state from:0 to:1 [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Announcer/ARI-00000031;2: [Aug 18 10:34:14] DEBUG[14444] channel.c: Channel Announcer/ARI-00000031;2 setting write format path: slin -> slin [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Announcer/ARI-00000031;2: [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Announcer/ARI-00000031;2: Not in SFU mode [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is joining softmix technology [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: SIP/zvonobot-0000003b: [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14448] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14447] res_rtp_asterisk.c: (0x7f0ca804b700) RTP changing ssrc from 126710661 to 2074252528 due to a source change [Aug 18 10:34:14] DEBUG[14419] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14419] http.c: HTTP closing session. Top level [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 From: ;tag=as04f0121c To: ;tag=as31dfa2de Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[14436] stasis.c: Topic 'cache:416/channel:213131': 0x7f0cac049550 created [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: SIP/zvonobot-0000003b: [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: SIP/zvonobot-0000003b: Not in SFU mode [Aug 18 10:34:14] DEBUG[14450] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is joining softmix technology [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Recorder/ARI-0000001e;2: [Aug 18 10:34:14] DEBUG[14444] channel.c: Channel Recorder/ARI-0000001e;2 setting write format path: slin -> slin [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:14] DEBUG[14444] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Recorder/ARI-0000001e;2: [Aug 18 10:34:14] DEBUG[14444] bridge_softmix.c: Recorder/ARI-0000001e;2: Not in SFU mode [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling softmix technology start [Aug 18 10:34:14] DEBUG[14444] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology destructor [Aug 18 10:34:14] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:14] DEBUG[14220] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14220] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:14] DEBUG[14450] http.c: HTTP Request URI is /ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/record?name=213036_CvDIuamjBInBGdVFxWOIOnQXrllKPTPz&format=wav [Aug 18 10:34:14] DEBUG[14418] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14418] chan_sip.c: SIP call-id changed from '27b89e867da32d886d1883b57c8eb1ec@127.0.1.1:5060' to '2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14418] stasis.c: Creating topic. name: channel:213125, detail: [Aug 18 10:34:14] DEBUG[14418] stasis.c: Topic 'channel:213125': 0x7f0c88052930 created [Aug 18 10:34:14] DEBUG[14418] stasis.c: Creating topic. name: cache:417/channel:213125, detail: [Aug 18 10:34:14] DEBUG[14418] stasis.c: Topic 'cache:417/channel:213125': 0x7f0c88055060 created [Aug 18 10:34:14] DEBUG[14450] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/record] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[14450] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/record] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as04f0121c [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31dfa2de [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: Allocating new SIP dialog for 34b651d6206463d6371d74063b4259d3@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14443] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c86070' [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) RTP allocated port 18670 [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE creating session 0.0.0.0:18670 (18670) [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE create [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE add system candidates [Aug 18 10:34:14] DEBUG[14443] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14443] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE add candidate: 159.65.48.104:18670, 2130706431 [Aug 18 10:34:14] DEBUG[14443] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14443] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE add candidate: 10.131.0.10:18670, 2130706431 [Aug 18 10:34:14] DEBUG[14443] rtp_engine.c: RTP instance '0x2c86070' is setup and ready to go [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) ICE stopped [Aug 18 10:34:14] DEBUG[14443] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14443] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14443] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14443] res_rtp_asterisk.c: (0x2c86070) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14443] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:14] DEBUG[14443] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Finding handler for channels/1629282827.33 [Aug 18 10:34:14] DEBUG[14450] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/record] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[14450] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pulling 0x7f0cac00a6f0(SIP/zvonobot-00000000) [Aug 18 10:34:14] VERBOSE[13056] bridge_channel.c: Channel SIP/zvonobot-00000000 left 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0cac00a6f0(SIP/zvonobot-00000000) is leaving simple_bridge technology [Aug 18 10:34:14] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Setting 0x7f0ca400f4f0(Recorder/ARI-00000000;2) state from:0 to:2 [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Finding handler for bridges/61075423-3ee2-4d60-8382-ee99e654a5be/record [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Finding handler for 1629282827.33 [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking channels create: Didn't match 1629282827.33 [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14448] res_ari.c: Checking channels externalMedia: Didn't match 1629282827.33 [Aug 18 10:34:14] DEBUG[14448] res_ari.c: No explicit handler found for 1629282827.33. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:14] DEBUG[14443] chan_sip.c: SIP call-id changed from '34b651d6206463d6371d74063b4259d3@127.0.1.1:5060' to '61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060' [Aug 18 10:34:14] DEBUG[14443] stasis.c: Creating topic. name: channel:213132, detail: [Aug 18 10:34:14] DEBUG[14443] stasis.c: Topic 'channel:213132': 0x2c2cb60 created [Aug 18 10:34:14] DEBUG[14443] stasis.c: Creating topic. name: cache:418/channel:213132, detail: [Aug 18 10:34:14] DEBUG[14443] stasis.c: Topic 'cache:418/channel:213132': 0x2c2fa80 created [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Finding handler for bridges [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:14] DEBUG[13920] chan_sip.c: Hangup call SIP/zvonobot-00000069, SIP callid 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[13920] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:14] DEBUG[13920] res_rtp_asterisk.c: (0x7f0c24122cb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13920] res_rtp_asterisk.c: (0x7f0c24122cb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13920] channel.c: Channel 0x7f0c24110ce0 'SIP/zvonobot-00000069' destroying [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:14] DEBUG[14216] stasis.c: Creating topic. name: channel:1629282854.363, detail: [Aug 18 10:34:14] DEBUG[14216] stasis.c: Topic 'channel:1629282854.363': 0x7f0c240761d0 created [Aug 18 10:34:14] DEBUG[14216] stasis.c: Creating topic. name: cache:419/channel:1629282854.363, detail: [Aug 18 10:34:14] DEBUG[14216] stasis.c: Topic 'cache:419/channel:1629282854.363': 0x7f0c241040e0 created [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282854.364, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.364': 0x7f0c300fba90 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: cache:420/channel:1629282854.364, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:420/channel:1629282854.364': 0x7f0c3005ada0 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: cache:420/channel:1629282854.364, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:420/channel:1629282854.364': 0x7f0c3005ada0 destroyed [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282854.364, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.364': 0x7f0c300fba90 destroyed [Aug 18 10:34:14] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000069', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213069', '')] [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel '213069': is 0 interested in calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel '213069' unsubscribed from calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: cache:265/channel:213069, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'cache:265/channel:213069': 0x7f0c240f9f70 destroyed [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: channel:213069, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'channel:213069': 0x7f0c240f94f0 destroyed [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 (Checking To) --From tag as04f0121c --To-tag as31dfa2de [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Stopping retransmission on '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae Max-Forwards: 70 From: ;tag=as04f0121c To: ;tag=as31dfa2de Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116945@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289 Max-Forwards: 70 From: ;tag=as09899d91 To: Contact: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 272107994 272107994 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116937@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b Max-Forwards: 70 From: ;tag=as17300792 To: Contact: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1659785286 1659785286 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (4) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (4) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #9 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #9)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116946@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786 Max-Forwards: 70 From: ;tag=as6230d06d To: Contact: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1405752376 1405752376 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19712 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000000 - start 1629282822.002824 answer 1629282827.019664 end 1629282854.650632 dur 32.647 bill 27.630 dispo ANSWERED [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (4) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (4) INVITE - 5 [Aug 18 10:34:14] DEBUG[14451] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14449] bridge_softmix.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: starting mixing thread [Aug 18 10:34:14] DEBUG[13996] res_stasis_playback.c: 1629282849.290: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:14] DEBUG[13056] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as two channels are required [Aug 18 10:34:14] DEBUG[13056] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:14] DEBUG[13056] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[13056] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[13056] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:14] DEBUG[13056] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c is already using the new technology. [Aug 18 10:34:14] DEBUG[14451] http.c: HTTP Request URI is /ari/channels/213069 [Aug 18 10:34:14] DEBUG[13996] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:14] DEBUG[13996] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Finding handler for 61075423-3ee2-4d60-8382-ee99e654a5be [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14453] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14450] res_ari.c: No explicit handler found for 61075423-3ee2-4d60-8382-ee99e654a5be. Using wildcard bridgeId. [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Finding handler for record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:14] DEBUG[14450] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:14] DEBUG[14450] stasis.c: Creating topic. name: channel:1629282854.365, detail: [Aug 18 10:34:14] DEBUG[14450] stasis.c: Topic 'channel:1629282854.365': 0x7f0c20048e90 created [Aug 18 10:34:14] DEBUG[14450] stasis.c: Creating topic. name: cache:421/channel:1629282854.365, detail: [Aug 18 10:34:14] DEBUG[14450] stasis.c: Topic 'cache:421/channel:1629282854.365': 0x7f0c200901b0 created [Aug 18 10:34:14] DEBUG[14451] http.c: match request [ari/channels/213069] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14453] http.c: HTTP Request URI is /ari/playbacks/483d16b8-74eb-44ec-8b57-6ca9cdb1a06e [Aug 18 10:34:14] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:14] DEBUG[14451] http.c: match request [ari/channels/213069] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[13058] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: pulling 0x7f0ca400f4f0(Recorder/ARI-00000000;2) [Aug 18 10:34:14] VERBOSE[13058] bridge_channel.c: Channel Recorder/ARI-00000000;2 left 'simple_bridge' stasis-bridge <7421ba4f-6229-4eeb-b806-91ebc84ff38c> [Aug 18 10:34:14] DEBUG[13058] bridge_channel.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: 0x7f0ca400f4f0(Recorder/ARI-00000000;2) is leaving simple_bridge technology [Aug 18 10:34:14] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTP audio difference is 55624, ms is 6973 [Aug 18 10:34:14] DEBUG[14451] http.c: match request [ari/channels/213069] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14451] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14453] http.c: match request [ari/playbacks/483d16b8-74eb-44ec-8b57-6ca9cdb1a06e] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[13058] bridge_native_rtp.c: Bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' can not use native RTP bridge as two channels are required [Aug 18 10:34:14] DEBUG[13058] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:14] DEBUG[13058] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14453] http.c: match request [ari/playbacks/483d16b8-74eb-44ec-8b57-6ca9cdb1a06e] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:14] DEBUG[13058] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[13058] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:14] DEBUG[13058] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c is already using the new technology. [Aug 18 10:34:14] DEBUG[14453] http.c: match request [ari/playbacks/483d16b8-74eb-44ec-8b57-6ca9cdb1a06e] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:34:14] DEBUG[13058] channel.c: Channel 0x7f0ca400e230 'Recorder/ARI-00000000;2' hanging up. Refs: 2 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14453] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Finding handler for playbacks/483d16b8-74eb-44ec-8b57-6ca9cdb1a06e [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Finding handler for playbacks [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Finding handler for channels/213069 [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Finding handler for channels [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6442ms with no response [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Hanging up call 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Finding handler for 213069 [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking channels create: Didn't match 213069 [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14451] res_ari.c: Checking channels externalMedia: Didn't match 213069 [Aug 18 10:34:14] DEBUG[14451] res_ari.c: No explicit handler found for 213069. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Bridge is returning 0x7f0cac00a6f0(SIP/zvonobot-00000000) to read format alaw [Aug 18 10:34:14] DEBUG[14218] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14218] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14160] channel.c: Channel 0x7f0c940b1960 'SIP/zvonobot-00000078' hanging up. Refs: 2 [Aug 18 10:34:14] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000078 - start 1629282847.730717 answer 0.000000 end 1629282854.746028 dur 7.015 bill 1629282854.746 dispo NO ANSWER [Aug 18 10:34:14] DEBUG[13550] audiohook.c: Audiohook 0x7f0ca8017670 has stale audio in its factories. Flushing them both [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:14] DEBUG[13056] channel.c: Channel SIP/zvonobot-00000000 setting read format path: alaw -> alaw [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:14] DEBUG[14169] channel.c: Channel 0x7f0c7008d490 'Snoop/212991-00000011' allocated [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Finding handler for 483d16b8-74eb-44ec-8b57-6ca9cdb1a06e [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[14453] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:14] DEBUG[13941] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b'. Checking compatability for channels 'SIP/zvonobot-0000001c' and 'Recorder/ARI-00000029;2' [Aug 18 10:34:14] DEBUG[13941] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as channel 'SIP/zvonobot-0000001c' has features which prevent it [Aug 18 10:34:14] DEBUG[13941] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:14] DEBUG[13941] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[13941] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:14] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 23 instead [Aug 18 10:34:14] DEBUG[13056] bridge_channel.c: Bridge is returning 0x7f0cac00a6f0(SIP/zvonobot-00000000) to write format alaw [Aug 18 10:34:14] DEBUG[14169] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:14] DEBUG[14169] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:14] DEBUG[14459] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[13941] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:14] DEBUG[14453] res_ari.c: No explicit handler found for 483d16b8-74eb-44ec-8b57-6ca9cdb1a06e. Using wildcard playbackId. [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:14] DEBUG[13056] channel.c: Channel SIP/zvonobot-00000000 setting write format path: alaw -> alaw [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:14] DEBUG[13941] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:14] DEBUG[14452] channel.c: Channel Announcer/ARI-00000031;1 setting write format path: gsm -> slin [Aug 18 10:34:14] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP got report of 100 bytes from 178.62.121.41:18113 [Aug 18 10:34:14] DEBUG[14453] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:14] DEBUG[14459] http.c: HTTP Request URI is /ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/play?media=sound%3Asilence%2F2 [Aug 18 10:34:14] DEBUG[14454] stasis/app.c: Channel '1629282851.322' is 1 interested in calls_0 [Aug 18 10:34:14] DEBUG[14454] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 145 instead [Aug 18 10:34:14] DEBUG[14453] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14463] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14462] http.c: HTTP opening session. Top level [Aug 18 10:34:14] DEBUG[14462] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212991&app=calls_0&format=slin16&external_host=127.0.0.1%3A50211 [Aug 18 10:34:14] DEBUG[14462] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14462] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[14459] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/play] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14462] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14463] http.c: HTTP Request URI is /ari/channels/robot_213023 [Aug 18 10:34:14] DEBUG[13056] stasis/control.c: 212964, 7421ba4f-6229-4eeb-b806-91ebc84ff38c: Channel was departed from bridge [Aug 18 10:34:14] DEBUG[13056] stasis/app.c: bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c': is 1 interested in calls_0 [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14459] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/play] with handler [phoneprov] len 9 [Aug 18 10:34:14] DEBUG[13056] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[12865] stasis/control.c: 212964: Channel departing bridge [Aug 18 10:34:14] DEBUG[14459] http.c: match request [ari/bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/play] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14462] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[12865] bridge.c: Waiting for 0x7f0cac00a6f0(SIP/zvonobot-00000000) bridge thread to die. [Aug 18 10:34:14] DEBUG[12865] stasis/app.c: channel '212964': is 1 interested in calls_0 [Aug 18 10:34:14] DEBUG[12865] channel.c: Channel 0x7f0cb0014a50 'SIP/zvonobot-00000000' hanging up. Refs: 3 [Aug 18 10:34:14] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 864, ms is 74 [Aug 18 10:34:14] DEBUG[14322] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:14] DEBUG[14459] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[14322] http.c: HTTP closing session. Top level [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Finding handler for bridges/5fd3583d-12a2-4028-9389-fce6801ffb6b/play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Finding handler for bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Finding handler for 5fd3583d-12a2-4028-9389-fce6801ffb6b [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14459] res_ari.c: No explicit handler found for 5fd3583d-12a2-4028-9389-fce6801ffb6b. Using wildcard bridgeId. [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Finding handler for play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:14] DEBUG[14459] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:14] DEBUG[14459] stasis.c: Creating topic. name: channel:1629282854.366, detail: [Aug 18 10:34:14] DEBUG[14459] stasis.c: Topic 'channel:1629282854.366': 0x7f0c3c13a4c0 created [Aug 18 10:34:14] DEBUG[14459] stasis.c: Creating topic. name: cache:422/channel:1629282854.366, detail: [Aug 18 10:34:14] DEBUG[14459] stasis.c: Topic 'cache:422/channel:1629282854.366': 0x7f0c3c13aef0 created [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[14463] http.c: match request [ari/channels/robot_213023] with handler [httpstatus] len 10 [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14462] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:14] DEBUG[14462] netsock2.c: Splitting '127.0.0.1:50211' into... [Aug 18 10:34:14] DEBUG[14462] netsock2.c: ...host '127.0.0.1' and port '50211'. [Aug 18 10:34:14] DEBUG[14462] netsock2.c: Splitting '127.0.0.1:50211' into... [Aug 18 10:34:14] DEBUG[14462] netsock2.c: ...host '127.0.0.1' and port '50211'. [Aug 18 10:34:14] DEBUG[14462] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:14] DEBUG[14462] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c38090cc0' [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) RTP allocated port 11252 [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE creating session 127.0.0.1:11252 (11252) [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE create [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:14] DEBUG[14452] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:14] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580;received=159.65.48.104 From: ;tag=as54647e8b To: ;tag=as194fe9f0 Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22fa0926" Content-Length: 0 <-------------> [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580;received=159.65.48.104 [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14463] http.c: match request [ari/channels/robot_213023] with handler [phoneprov] len 9 [Aug 18 10:34:14] VERBOSE[14452] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:14] DEBUG[14463] http.c: match request [ari/channels/robot_213023] with handler [ari] len 3 [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE add system candidates [Aug 18 10:34:14] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54647e8b [Aug 18 10:34:14] DEBUG[14463] http.c: Match made with [ari] [Aug 18 10:34:14] DEBUG[13497] res_rtp_asterisk.c: (0x7f0c080871b0) DTLS stop [Aug 18 10:34:14] DEBUG[13497] res_rtp_asterisk.c: (0x7f0c080871b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:14] DEBUG[13497] res_rtp_asterisk.c: (0x7f0c080871b0) ICE RTP transport deallocating [Aug 18 10:34:14] DEBUG[13497] res_rtp_asterisk.c: (0x7f0c080871b0) ICE stopped [Aug 18 10:34:14] DEBUG[13497] rtp_engine.c: Destroyed RTP instance '0x7f0c080871b0' [Aug 18 10:34:14] DEBUG[13497] channel.c: Channel 0x7f0c08053190 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0' destroying [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as194fe9f0 [Aug 18 10:34:14] DEBUG[14340] stasis.c: Creating topic. name: channel:1629282854.367, detail: [Aug 18 10:34:14] DEBUG[14340] stasis.c: Topic 'channel:1629282854.367': 0x7f0c3005a870 created [Aug 18 10:34:14] DEBUG[14340] stasis.c: Creating topic. name: cache:423/channel:1629282854.367, detail: [Aug 18 10:34:14] DEBUG[14340] stasis.c: Topic 'cache:423/channel:1629282854.367': 0x7f0c300e56d0 created [Aug 18 10:34:14] DEBUG[14462] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282854.368, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.368': 0x7f0c30108360 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Creating topic. name: cache:424/channel:1629282854.368, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:424/channel:1629282854.368': 0x7f0c3011b7d0 created [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: cache:424/channel:1629282854.368, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'cache:424/channel:1629282854.368': 0x7f0c3011b7d0 destroyed [Aug 18 10:34:14] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282854.368, detail: [Aug 18 10:34:14] DEBUG[20545] stasis.c: Topic 'channel:1629282854.368': 0x7f0c30108360 destroyed [Aug 18 10:34:14] DEBUG[14462] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[13327] audiohook.c: Audiohook 0x7f0c9c02ce30 has stale audio in its factories. Flushing them both [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:14] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:55', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50433-0x7f0c080871b0', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213012', '')] [Aug 18 10:34:14] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel 'robot_213012': is 0 interested in calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis/app.c: channel 'robot_213012' unsubscribed from calls_0 [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: cache:162/channel:robot_213012, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'cache:162/channel:robot_213012': 0x7f0c08055370 destroyed [Aug 18 10:34:14] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213012, detail: [Aug 18 10:34:14] DEBUG[20620] stasis.c: Topic 'channel:robot_213012': 0x7f0c08050ee0 destroyed [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Finding handler for channels/robot_213023 [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE add candidate: 159.65.48.104:11252, 2130706431 [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Finding handler for channels [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22fa0926" [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 (Checking To) --From tag as54647e8b --To-tag as194fe9f0 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (3) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #6 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #6)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116938@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0 Max-Forwards: 70 From: ;tag=as2e1ef431 To: Contact: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 424817061 424817061 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14986 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #94 (5) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #94)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116939@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68796b76 Max-Forwards: 70 From: ;tag=as7ed89ca7 To: Contact: Call-ID: 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1442145806 1442145806 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (4) INVITE - 5 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:14] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:14] DEBUG[14462] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14462] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:14] DEBUG[14351] stasis.c: Creating topic. name: channel:1629282854.369, detail: [Aug 18 10:34:14] DEBUG[14351] stasis.c: Topic 'channel:1629282854.369': 0x7f0c70013ed0 created [Aug 18 10:34:14] DEBUG[14351] stasis.c: Creating topic. name: cache:425/channel:1629282854.369, detail: [Aug 18 10:34:14] DEBUG[14351] stasis.c: Topic 'cache:425/channel:1629282854.369': 0x7f0c7005b290 created [Aug 18 10:34:14] DEBUG[14440] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:14] DEBUG[14440] chan_sip.c: Allocating new SIP dialog for 2f76961a194703790f3ae1c00cedad3c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Finding handler for robot_213023 [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6271ms with no response [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking channels create: Didn't match robot_213023 [Aug 18 10:34:14] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:14] DEBUG[14440] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4045900' [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) RTP allocated port 13424 [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE creating session 0.0.0.0:13424 (13424) [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE create [Aug 18 10:34:14] WARNING[20585] chan_sip.c: Hanging up call 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add system candidates [Aug 18 10:34:14] DEBUG[14463] res_ari.c: Checking channels externalMedia: Didn't match robot_213023 [Aug 18 10:34:14] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:14] DEBUG[14440] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:14] DEBUG[14440] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add candidate: 159.65.48.104:13424, 2130706431 [Aug 18 10:34:14] DEBUG[14440] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:14] DEBUG[14440] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE add candidate: 10.131.0.10:13424, 2130706431 [Aug 18 10:34:14] DEBUG[14440] rtp_engine.c: RTP instance '0x7f0cb4045900' is setup and ready to go [Aug 18 10:34:14] DEBUG[14463] res_ari.c: No explicit handler found for robot_213023. Using wildcard channelId. [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) ICE stopped [Aug 18 10:34:14] DEBUG[14440] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:14] DEBUG[14440] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:14] DEBUG[14440] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:14] DEBUG[14440] res_rtp_asterisk.c: (0x7f0cb4045900) RTCP setup on RTP instance [Aug 18 10:34:14] VERBOSE[14440] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:14] DEBUG[14462] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE add candidate: 10.131.0.10:11252, 2130706431 [Aug 18 10:34:14] DEBUG[14440] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:15] DEBUG[14440] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:15] DEBUG[14440] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:15] DEBUG[14440] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14440] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:15] DEBUG[13187] res_rtp_asterisk.c: (0x7f0c8403cbb0) RTP 0x7f0c84042730 -- Received packet from 178.62.121.41:13958, dropping due to strict RTP protection. [Aug 18 10:34:15] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:15] DEBUG[14462] rtp_engine.c: RTP instance '0x7f0c38090cc0' is setup and ready to go [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14189] channel.c: Channel 0x7f0ca80ecaf0 'SIP/zvonobot-0000007b' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007b - start 1629282848.380933 answer 0.000000 end 1629282855.022330 dur 6.641 bill 1629282855.022 dispo NO ANSWER [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:15] DEBUG[14440] chan_sip.c: SIP call-id changed from '2f76961a194703790f3ae1c00cedad3c@127.0.1.1:5060' to '2b2796597f47b4537f2a70003282d682@159.65.48.104:5060' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24122cb0) DTLS stop [Aug 18 10:34:15] DEBUG[14440] stasis.c: Creating topic. name: channel:213133, detail: [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24122cb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14440] stasis.c: Topic 'channel:213133': 0x7f0cb40362d0 created [Aug 18 10:34:15] DEBUG[14452] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:15] DEBUG[14440] stasis.c: Creating topic. name: cache:426/channel:213133, detail: [Aug 18 10:34:15] DEBUG[14039] chan_sip.c: Hangup call SIP/zvonobot-0000006e, SIP callid 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14039] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14039] res_rtp_asterisk.c: (0x7f0c9408df40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14039] res_rtp_asterisk.c: (0x7f0c9408df40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14039] channel.c: Channel 0x7f0c9409b680 'SIP/zvonobot-0000006e' destroying [Aug 18 10:34:15] DEBUG[14035] chan_sip.c: Hangup call SIP/zvonobot-00000068, SIP callid 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14035] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14035] res_rtp_asterisk.c: (0x7f0c4005ac00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14035] res_rtp_asterisk.c: (0x7f0c4005ac00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14035] channel.c: Channel 0x7f0c400470e0 'SIP/zvonobot-00000068' destroying [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213076': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213076' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.372, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.372': 0x7f0c30108360 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:427/channel:1629282855.372, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:427/channel:1629282855.372': 0x7f0c3011b7d0 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:427/channel:1629282855.372, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:427/channel:1629282855.372': 0x7f0c3011b7d0 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.372, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.372': 0x7f0c30108360 destroyed [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213076', '')] [Aug 18 10:34:15] DEBUG[14462] stasis.c: Creating topic. name: channel:robot_212991, detail: [Aug 18 10:34:15] DEBUG[14440] stasis.c: Topic 'cache:426/channel:213133': 0x7f0cb4007560 created [Aug 18 10:34:15] DEBUG[14464] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[14452] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[14336] channel.c: Channel 0x7f0c280d8530 'SIP/zvonobot-00000097' allocated [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14336] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24122cb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14464] http.c: HTTP Request URI is /ari/channels/213076 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.373, detail: [Aug 18 10:34:15] DEBUG[14464] http.c: match request [ari/channels/213076] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213073': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.373': 0x7f0c3005e390 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:428/channel:1629282855.373, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:428/channel:1629282855.373': 0x7f0c301203e0 created [Aug 18 10:34:15] DEBUG[14464] http.c: match request [ari/channels/213076] with handler [phoneprov] len 9 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213073' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:15] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:15] DEBUG[14464] http.c: match request [ari/channels/213076] with handler [ari] len 3 [Aug 18 10:34:15] DEBUG[14464] http.c: Match made with [ari] [Aug 18 10:34:15] DEBUG[14039] stasis.c: Destroying topic. name: cache:280/channel:213076, detail: [Aug 18 10:34:15] DEBUG[14039] stasis.c: Topic 'cache:280/channel:213076': 0x7f0c9409de80 destroyed [Aug 18 10:34:15] DEBUG[14039] stasis.c: Destroying topic. name: channel:213076, detail: [Aug 18 10:34:15] DEBUG[14039] stasis.c: Topic 'channel:213076': 0x7f0c9409d400 destroyed [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Finding handler for channels/213076 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:428/channel:1629282855.373, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:428/channel:1629282855.373': 0x7f0c301203e0 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.373, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.373': 0x7f0c3005e390 destroyed [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000068', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213073', '')] [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: cache:263/channel:213073, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'cache:263/channel:213073': 0x7f0c400498e0 destroyed [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: channel:213073, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'channel:213073': 0x7f0c40048e60 destroyed [Aug 18 10:34:15] DEBUG[14465] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24122cb0) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14452] channel.c: Channel Announcer/ARI-00000031;1 setting write format path: slin -> slin [Aug 18 10:34:15] NOTICE[14452] res_stasis_playback.c: 1629282849.290: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c24122cb0' [Aug 18 10:34:15] DEBUG[14465] http.c: HTTP Request URI is /ari/channels/213073 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[14462] stasis.c: Topic 'channel:robot_212991': 0x7f0c3807e8f0 created [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Finding handler for 213076 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking channels create: Didn't match 213076 [Aug 18 10:34:15] DEBUG[14465] http.c: match request [ari/channels/213073] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:15] DEBUG[14462] stasis.c: Creating topic. name: cache:429/channel:robot_212991, detail: [Aug 18 10:34:15] DEBUG[14464] res_ari.c: Checking channels externalMedia: Didn't match 213076 [Aug 18 10:34:15] DEBUG[14464] res_ari.c: No explicit handler found for 213076. Using wildcard channelId. [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6391ms with no response [Aug 18 10:34:15] DEBUG[14465] http.c: match request [ari/channels/213073] with handler [phoneprov] len 9 [Aug 18 10:34:15] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:15] DEBUG[14465] http.c: match request [ari/channels/213073] with handler [ari] len 3 [Aug 18 10:34:15] DEBUG[14465] http.c: Match made with [ari] [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Hanging up call 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14452] channel.c: Channel 0x7f0c2001ad30 'Announcer/ARI-00000031;1' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[14462] stasis.c: Topic 'cache:429/channel:robot_212991': 0x7f0c38066b90 created [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Finding handler for channels/213073 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14048] chan_sip.c: Hangup call SIP/zvonobot-00000071, SIP callid 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[14048] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14048] res_rtp_asterisk.c: (0x7f0c8c0381d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14048] res_rtp_asterisk.c: (0x7f0c8c0381d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14048] channel.c: Channel 0x7f0c8c050630 'SIP/zvonobot-00000071' destroying [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.374, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.374': 0x7f0c30108360 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:430/channel:1629282855.374, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:430/channel:1629282855.374': 0x7f0c300b35e0 created [Aug 18 10:34:15] DEBUG[14190] channel.c: Channel 0x7f0ca4057750 'SIP/zvonobot-0000007c' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[14339] channel.c: Channel 0x7f0c340f0030 'SIP/zvonobot-00000098' allocated [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[14336] res_stasis.c: calls_0: Subscribing to 213116 [Aug 18 10:34:15] DEBUG[14336] stasis/app.c: Channel '213116' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Outgoing Call for 79821116924 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14336] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213074': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213074' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:15] DEBUG[14336] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14467] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[14467] http.c: HTTP Request URI is /ari/channels/213074 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Finding handler for 213073 [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14339] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking channels create: Didn't match 213073 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[14465] res_ari.c: Checking channels externalMedia: Didn't match 213073 [Aug 18 10:34:15] DEBUG[14465] res_ari.c: No explicit handler found for 213073. Using wildcard channelId. [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:15] DEBUG[14467] http.c: match request [ari/channels/213074] with handler [httpstatus] len 10 [Aug 18 10:34:15] VERBOSE[14466] chan_sip.c: Audio is at 12662 [Aug 18 10:34:15] VERBOSE[14466] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] DEBUG[14467] http.c: match request [ari/channels/213074] with handler [phoneprov] len 9 [Aug 18 10:34:15] VERBOSE[14466] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] VERBOSE[14466] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14339] res_stasis.c: calls_0: Subscribing to 213115 [Aug 18 10:34:15] DEBUG[14467] http.c: match request [ari/channels/213074] with handler [ari] len 3 [Aug 18 10:34:15] DEBUG[14339] stasis/app.c: Channel '213115' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14339] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14339] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Outgoing Call for 79821116925 [Aug 18 10:34:15] DEBUG[14048] stasis.c: Destroying topic. name: cache:283/channel:213074, detail: [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14048] stasis.c: Topic 'cache:283/channel:213074': 0x7f0c8c0e7690 destroyed [Aug 18 10:34:15] DEBUG[14048] stasis.c: Destroying topic. name: channel:213074, detail: [Aug 18 10:34:15] DEBUG[14048] stasis.c: Topic 'channel:213074': 0x7f0c8c05c950 destroyed [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14468] chan_sip.c: Audio is at 17556 [Aug 18 10:34:15] VERBOSE[14468] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] DEBUG[14467] http.c: Match made with [ari] [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:430/channel:1629282855.374, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:430/channel:1629282855.374': 0x7f0c300b35e0 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.374, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.374': 0x7f0c30108360 destroyed [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000071', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213074', '')] [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Initializing initreq for method INVITE - callid 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116924@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] VERBOSE[14468] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db [Aug 18 10:34:15] VERBOSE[14468] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 3 [ 52]: From: ;tag=as1e86ce5f [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 6 [ 60]: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14466] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #109 [Aug 18 10:34:15] DEBUG[14466] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14352] channel.c: Channel 0x7f0c7c0b37d0 'SIP/zvonobot-0000009a' allocated [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14352] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] VERBOSE[14466] dial.c: Called zvonobot/79821116924 [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Initializing initreq for method INVITE - callid 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007c - start 1629282848.467696 answer 0.000000 end 1629282855.207069 dur 6.739 bill 1629282855.207 dispo NO ANSWER [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Finding handler for channels/213074 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[14352] res_stasis.c: calls_0: Subscribing to 213118 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda;received=159.65.48.104 From: ;tag=as5b87d923 To: ;tag=as5382feb7 Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68236f1d" Content-Length: 0 <-------------> [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116925@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 [Aug 18 10:34:15] DEBUG[14352] stasis/app.c: Channel '213118' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK558c8cda;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Finding handler for 213074 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking channels create: Didn't match 213074 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[14467] res_ari.c: Checking channels externalMedia: Didn't match 213074 [Aug 18 10:34:15] DEBUG[14467] res_ari.c: No explicit handler found for 213074. Using wildcard channelId. [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14352] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 3 [ 52]: From: ;tag=as0b888cfb [Aug 18 10:34:15] DEBUG[14352] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Outgoing Call for 79821116922 [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5b87d923 [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 6 [ 60]: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5382feb7 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14468] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116925@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 Max-Forwards: 70 From: ;tag=as0b888cfb To: Contact: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 211868227 211868227 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #86 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[14468] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] VERBOSE[14468] dial.c: Called zvonobot/79821116925 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68236f1d" [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:15] VERBOSE[14471] chan_sip.c: Audio is at 12284 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 (Checking To) --From tag as5b87d923 --To-tag as5382feb7 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2c5322d560d5755f39711b55002aec77@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:15] VERBOSE[14471] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] VERBOSE[14471] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (5) INVITE - 5 [Aug 18 10:34:15] VERBOSE[14471] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116941@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd Max-Forwards: 70 From: ;tag=as3f0bc324 To: Contact: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1205493209 1205493209 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10796 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Initializing initreq for method INVITE - callid 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (4) INVITE - 5 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116922@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 3 [ 52]: From: ;tag=as7674a2b1 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 6 [ 60]: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (4) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTCP got report of 100 bytes from 178.62.121.41:10913 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (4) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (4) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14471] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14471] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] VERBOSE[14471] dial.c: Called zvonobot/79821116922 [Aug 18 10:34:15] DEBUG[14050] chan_sip.c: Hangup call SIP/zvonobot-00000070, SIP callid 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14050] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14050] res_rtp_asterisk.c: (0x7f0c9804b2b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14050] res_rtp_asterisk.c: (0x7f0c9804b2b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14050] channel.c: Channel 0x7f0c9803da40 'SIP/zvonobot-00000070' destroying [Aug 18 10:34:15] DEBUG[14342] channel.c: Channel 0x7f0c3c1318b0 'SIP/zvonobot-00000099' allocated [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14342] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '0c08b1570fa732272364833678dc04bb@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408df40) DTLS stop [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408df40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213078': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213078' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408df40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.375, detail: [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408df40) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9408df40' [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5993fccb0e95740465a028667804b469@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6454ms with no response [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Hanging up call 5993fccb0e95740465a028667804b469@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.375': 0x7f0c30108360 created [Aug 18 10:34:15] DEBUG[14474] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14474] http.c: HTTP Request URI is /ari/channels/213078 [Aug 18 10:34:15] DEBUG[14193] channel.c: Channel 0x7f0c98081820 'SIP/zvonobot-0000007d' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[14050] stasis.c: Destroying topic. name: cache:282/channel:213078, detail: [Aug 18 10:34:15] DEBUG[14050] stasis.c: Topic 'cache:282/channel:213078': 0x7f0c98040240 destroyed [Aug 18 10:34:15] DEBUG[14050] stasis.c: Destroying topic. name: channel:213078, detail: [Aug 18 10:34:15] DEBUG[14050] stasis.c: Topic 'channel:213078': 0x7f0c9803f7c0 destroyed [Aug 18 10:34:15] DEBUG[14474] http.c: match request [ari/channels/213078] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[14474] http.c: match request [ari/channels/213078] with handler [phoneprov] len 9 [Aug 18 10:34:15] DEBUG[14474] http.c: match request [ari/channels/213078] with handler [ari] len 3 [Aug 18 10:34:15] DEBUG[14474] http.c: Match made with [ari] [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Finding handler for channels/213078 [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Finding handler for 213078 [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking channels create: Didn't match 213078 [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[14474] res_ari.c: Checking channels externalMedia: Didn't match 213078 [Aug 18 10:34:15] DEBUG[14474] res_ari.c: No explicit handler found for 213078. Using wildcard channelId. [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:431/channel:1629282855.375, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:431/channel:1629282855.375': 0x7f0c300fb920 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:431/channel:1629282855.375, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:431/channel:1629282855.375': 0x7f0c300fb920 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.375, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.375': 0x7f0c30108360 destroyed [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000070', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213078', '')] [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:15] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007d - start 1629282848.710691 answer 0.000000 end 1629282855.459409 dur 6.748 bill 1629282855.459 dispo NO ANSWER [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4005ac00) DTLS stop [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4005ac00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4005ac00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4005ac00) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c4005ac00' [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '42462bcf58720fbb2059b6de455547db@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0381d0) DTLS stop [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0381d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0381d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0381d0) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8c0381d0' [Aug 18 10:34:15] DEBUG[14342] res_stasis.c: calls_0: Subscribing to 213117 [Aug 18 10:34:15] DEBUG[14342] stasis/app.c: Channel '213117' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Outgoing Call for 79821116923 [Aug 18 10:34:15] DEBUG[14342] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14342] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14475] chan_sip.c: Audio is at 18958 [Aug 18 10:34:15] VERBOSE[14475] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] VERBOSE[14475] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] VERBOSE[14475] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Initializing initreq for method INVITE - callid 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116923@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734;received=159.65.48.104 From: ;tag=as611ff9f7 To: ;tag=as0807854b Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51d63101" Content-Length: 0 <-------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as611ff9f7 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0807854b [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51d63101" [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 (Checking To) --From tag as611ff9f7 --To-tag as0807854b [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (1) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 3 [ 52]: From: ;tag=as6400b9b5 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 6 [ 60]: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (1) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116925@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 Max-Forwards: 70 From: ;tag=as0b888cfb To: Contact: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 211868227 211868227 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14475] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 343394319 343394319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18958 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #79 [Aug 18 10:34:15] DEBUG[14475] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14376] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:15] DEBUG[14376] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (4) INVITE - 5 [Aug 18 10:34:15] VERBOSE[14475] dial.c: Called zvonobot/79821116923 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6327ms with no response [Aug 18 10:34:15] WARNING[20585] chan_sip.c: Hanging up call 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] DEBUG[14202] channel.c: Channel 0x7f0cb4073e80 'SIP/zvonobot-0000007f' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007f - start 1629282849.087721 answer 0.000000 end 1629282855.549557 dur 6.461 bill 1629282855.549 dispo NO ANSWER [Aug 18 10:34:15] DEBUG[14354] channel.c: Channel 0x7f0c84094380 'SIP/zvonobot-0000009b' allocated [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14354] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[14368] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:15] DEBUG[14368] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9804b2b0) DTLS stop [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9804b2b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9804b2b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9804b2b0) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9804b2b0' [Aug 18 10:34:15] DEBUG[14090] chan_sip.c: Hangup call SIP/zvonobot-00000073, SIP callid 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14090] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14090] res_rtp_asterisk.c: (0x7f0ca402ac60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14090] res_rtp_asterisk.c: (0x7f0ca402ac60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14090] channel.c: Channel 0x7f0ca4040e00 'SIP/zvonobot-00000073' destroying [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.376, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.376': 0x7f0c300e65a0 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:432/channel:1629282855.376, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:432/channel:1629282855.376': 0x7f0c3005e390 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:432/channel:1629282855.376, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:432/channel:1629282855.376': 0x7f0c3005e390 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.376, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.376': 0x7f0c300e65a0 destroyed [Aug 18 10:34:15] DEBUG[14353] channel.c: Channel 0x7f0c78022bf0 'SIP/zvonobot-0000009c' allocated [Aug 18 10:34:15] DEBUG[14354] res_stasis.c: calls_0: Subscribing to 213122 [Aug 18 10:34:15] DEBUG[14354] stasis/app.c: Channel '213122' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213082': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213082' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: cache:285/channel:213082, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'cache:285/channel:213082': 0x7f0ca4043600 destroyed [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: channel:213082, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'channel:213082': 0x7f0ca4042b80 destroyed [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:06', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000073', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213082', '')] [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2f2b4180 Max-Forwards: 70 From: ;tag=as6e8cb5a3 To: ;tag=as686a751a Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="2f11f39e", response="346067b717392833ece8dfb2e8759785" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2f2b4180 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6e8cb5a3 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as686a751a [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="2f11f39e", response="346067b717392833ece8dfb2e8759785" [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 (Checking From) --From tag as6e8cb5a3 --To-tag as686a751a [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:15] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Outgoing Call for 79821116918 [Aug 18 10:34:15] DEBUG[14354] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14478] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[14354] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14477] chan_sip.c: Audio is at 12398 [Aug 18 10:34:15] VERBOSE[14477] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] VERBOSE[14477] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] VERBOSE[14477] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Initializing initreq for method INVITE - callid 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116918@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 3 [ 52]: From: ;tag=as2a62315e [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 6 [ 60]: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14477] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #104 [Aug 18 10:34:15] DEBUG[14477] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[14477] dial.c: Called zvonobot/79821116918 [Aug 18 10:34:15] DEBUG[14478] http.c: HTTP Request URI is /ari/channels/213082 [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14353] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '398559732fb8625271bea90231b90490@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2f2b4180;received=178.62.121.41 From: ;tag=as6e8cb5a3 To: ;tag=as686a751a Call-ID: 398559732fb8625271bea90231b90490@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14478] http.c: match request [ari/channels/213082] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (1) INVITE - 5 [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Setting 0x7f0c7006de00(SIP/zvonobot-0000002a) state from:0 to:1 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pulling 0x7f0c7006de00(SIP/zvonobot-0000002a) [Aug 18 10:34:15] DEBUG[14478] http.c: match request [ari/channels/213082] with handler [phoneprov] len 9 [Aug 18 10:34:15] VERBOSE[13666] bridge_channel.c: Channel SIP/zvonobot-0000002a left 'simple_bridge' stasis-bridge [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c7006de00(SIP/zvonobot-0000002a) is leaving simple_bridge technology [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Setting 0x7f0c2c08b700(Recorder/ARI-00000020;2) state from:0 to:2 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (2) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14353] res_stasis.c: calls_0: Subscribing to 213119 [Aug 18 10:34:15] DEBUG[14353] stasis/app.c: Channel '213119' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14478] http.c: match request [ari/channels/213082] with handler [ari] len 3 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:15] DEBUG[14478] http.c: Match made with [ari] [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Outgoing Call for 79821116921 [Aug 18 10:34:15] DEBUG[14353] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14353] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[13666] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as two channels are required [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca402ac60) DTLS stop [Aug 18 10:34:15] DEBUG[13666] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca402ac60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca402ac60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca402ac60) ICE RTP transport deallocating [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca402ac60' [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Session timer stopped: 54 - 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (1) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 343394319 343394319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18958 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787;received=159.65.48.104 From: ;tag=as5a5dd50f To: ;tag=as7e9bd570 Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78230a22" Content-Length: 0 <-------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK443f7787;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5a5dd50f [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7e9bd570 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78230a22" [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 (Checking To) --From tag as5a5dd50f --To-tag as7e9bd570 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 25cf37db0013b0ab0e87b7b45cf0b712@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Finding handler for channels/213082 [Aug 18 10:34:15] DEBUG[13666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[13666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[13666] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:15] DEBUG[13666] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[13702] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pulling 0x7f0c2c08b700(Recorder/ARI-00000020;2) [Aug 18 10:34:15] VERBOSE[13702] bridge_channel.c: Channel Recorder/ARI-00000020;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[13702] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c2c08b700(Recorder/ARI-00000020;2) is leaving simple_bridge technology [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Finding handler for 213082 [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking channels create: Didn't match 213082 [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Bridge is returning 0x7f0c7006de00(SIP/zvonobot-0000002a) to read format alaw [Aug 18 10:34:15] DEBUG[13666] channel.c: Channel SIP/zvonobot-0000002a setting read format path: ulaw -> alaw [Aug 18 10:34:15] DEBUG[13666] bridge_channel.c: Bridge is returning 0x7f0c7006de00(SIP/zvonobot-0000002a) to write format alaw [Aug 18 10:34:15] DEBUG[13666] channel.c: Channel SIP/zvonobot-0000002a setting write format path: alaw -> ulaw [Aug 18 10:34:15] DEBUG[13666] stasis/control.c: 213007, aba705f1-c39f-408a-8a02-8c7f66ee7c7d: Channel was departed from bridge [Aug 18 10:34:15] DEBUG[13666] stasis/app.c: bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d': is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[13666] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:15] DEBUG[13177] stasis/control.c: 213007: Channel departing bridge [Aug 18 10:34:15] DEBUG[13177] bridge.c: Waiting for 0x7f0c7006de00(SIP/zvonobot-0000002a) bridge thread to die. [Aug 18 10:34:15] DEBUG[13177] stasis/app.c: channel '213007': is 1 interested in calls_0 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 From: ;tag=as009e460a To: ;tag=as64de9d5c Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 568221000 568221000 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13198 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:15] DEBUG[13177] channel.c: Channel 0x7f0c74015470 'SIP/zvonobot-0000002a' hanging up. Refs: 3 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5ce4c6ac;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[14478] res_ari.c: Checking channels externalMedia: Didn't match 213082 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as009e460a [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:15] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000002a - start 1629282830.069057 answer 1629282839.260639 end 1629282855.651280 dur 25.582 bill 16.390 dispo ANSWERED [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] DEBUG[14478] res_ari.c: No explicit handler found for 213082. Using wildcard channelId. [Aug 18 10:34:15] DEBUG[13702] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as two channels are required [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64de9d5c [Aug 18 10:34:15] DEBUG[13702] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:15] DEBUG[13702] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[13702] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[13702] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:15] DEBUG[13702] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:15] VERBOSE[14479] chan_sip.c: Audio is at 15158 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:15] DEBUG[13702] channel.c: Channel 0x7f0c2c096fd0 'Recorder/ARI-00000020;2' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 568221000 568221000 IN IP4 178.62.121.41 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13198 RTP/AVP 0 8 101 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:15] VERBOSE[14479] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 (Checking To) --From tag as009e460a --To-tag as64de9d5c [Aug 18 10:34:15] DEBUG[14349] app.c: One waitfor failed, trying another [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Stopping retransmission on '2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:15] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:15] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117018@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0bbaf1f3 Max-Forwards: 70 From: ;tag=as009e460a To: ;tag=as64de9d5c Contact: Call-ID: 2cfac4f83d95cff054e5ebfd1f31c102@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[14479] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (2) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116925@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 Max-Forwards: 70 From: ;tag=as0b888cfb To: Contact: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 211868227 211868227 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (6) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116950@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573 Max-Forwards: 70 From: ;tag=as05f1fb09 To: Contact: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1424807793 1424807793 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17384 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (1) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (2) INVITE - 5 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[14479] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Initializing initreq for method INVITE - callid 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a;received=159.65.48.104 From: ;tag=as1ed67fff To: ;tag=as2fcf303b Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1132db38" Content-Length: 0 <-------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK45fdc18a;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116921@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1ed67fff [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 3 [ 52]: From: ;tag=as41c9ab07 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2fcf303b [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 6 [ 60]: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1132db38" [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 (Checking To) --From tag as1ed67fff --To-tag as2fcf303b [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 43b4e25c08fffda709b216ba0f67f7d3@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14479] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #95 [Aug 18 10:34:15] DEBUG[14479] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[14279] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:15] DEBUG[14279] http.c: HTTP closing session. Top level [Aug 18 10:34:15] VERBOSE[14479] dial.c: Called zvonobot/79821116921 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: ;tag=as6089b55c Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1145089429 1145089429 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12380 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6089b55c [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14359] channel.c: Channel 0x7f0c940ed7b0 'SIP/zvonobot-0000009d' allocated [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:15] DEBUG[14359] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:15] DEBUG[14117] chan_sip.c: Hangup call SIP/zvonobot-00000074, SIP callid 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14117] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14117] res_rtp_asterisk.c: (0x7f0cac077690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14156] channel.c: Channel 0x7f0c78090610 'Recorder/ARI-00000029;1' destroying [Aug 18 10:34:15] DEBUG[14117] res_rtp_asterisk.c: (0x7f0cac077690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14117] channel.c: Channel 0x7f0cac095830 'SIP/zvonobot-00000074' destroying [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1145089429 1145089429 IN IP4 178.62.121.41 [Aug 18 10:34:15] DEBUG[13741] channel.c: Channel 0x7f0c2c08ce90 'Recorder/ARI-00000020;1' destroying [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:15] DEBUG[14143] bridge_channel.c: Setting 0x7f0c7804b2f0(Recorder/ARI-00000029;2) state from:0 to:1 [Aug 18 10:34:15] DEBUG[14347] channel.c: Channel 0x7f0c38004d20 'SIP/zvonobot-0000009e' allocated [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14347] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213081': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213081' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: cache:287/channel:213081, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'cache:287/channel:213081': 0x7f0cac098340 destroyed [Aug 18 10:34:15] DEBUG[20620] stasis.c: Destroying topic. name: channel:213081, detail: [Aug 18 10:34:15] DEBUG[20620] stasis.c: Topic 'channel:213081': 0x7f0cac097940 destroyed [Aug 18 10:34:15] DEBUG[14156] stasis.c: Destroying topic. name: cache:279/channel:1629282843.237, detail: [Aug 18 10:34:15] DEBUG[14156] stasis.c: Topic 'cache:279/channel:1629282843.237': 0x7f0c78040590 destroyed [Aug 18 10:34:15] DEBUG[14156] stasis.c: Destroying topic. name: channel:1629282843.237, detail: [Aug 18 10:34:15] DEBUG[13741] stasis.c: Destroying topic. name: cache:216/channel:1629282839.181, detail: [Aug 18 10:34:15] DEBUG[13741] stasis.c: Topic 'cache:216/channel:1629282839.181': 0x7f0c2c008a60 destroyed [Aug 18 10:34:15] DEBUG[13741] stasis.c: Destroying topic. name: channel:1629282839.181, detail: [Aug 18 10:34:15] DEBUG[13741] stasis.c: Topic 'channel:1629282839.181': 0x7f0c2c07fd10 destroyed [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.377, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.377': 0x7f0c300b3540 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:433/channel:1629282855.377, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:433/channel:1629282855.377': 0x7f0c3011cb70 created [Aug 18 10:34:15] DEBUG[14143] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pulling 0x7f0c7804b2f0(Recorder/ARI-00000029;2) [Aug 18 10:34:15] VERBOSE[14143] bridge_channel.c: Channel Recorder/ARI-00000029;2 left 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:15] DEBUG[14143] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c7804b2f0(Recorder/ARI-00000029;2) is leaving simple_bridge technology [Aug 18 10:34:15] DEBUG[14482] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[14156] stasis.c: Topic 'channel:1629282843.237': 0x7f0c7806efc0 destroyed [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12380 RTP/AVP 0 8 101 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:15] DEBUG[14482] http.c: HTTP Request URI is /ari/channels/213081 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:433/channel:1629282855.377, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:433/channel:1629282855.377': 0x7f0c3011cb70 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.377, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.377': 0x7f0c300b3540 destroyed [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:06', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000074', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213081', '')] [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:15] DEBUG[14482] http.c: match request [ari/channels/213081] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[14482] http.c: match request [ari/channels/213081] with handler [phoneprov] len 9 [Aug 18 10:34:15] DEBUG[14482] http.c: match request [ari/channels/213081] with handler [ari] len 3 [Aug 18 10:34:15] DEBUG[14482] http.c: Match made with [ari] [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:15] DEBUG[14143] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Finding handler for channels/213081 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:15] DEBUG[14143] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:15] DEBUG[14143] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag as6089b55c [Aug 18 10:34:15] DEBUG[14143] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:15] DEBUG[14359] res_stasis.c: calls_0: Subscribing to 213123 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:15] DEBUG[14359] stasis/app.c: Channel '213123' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14143] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Stopping retransmission on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:15] DEBUG[14143] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:15] DEBUG[14359] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14359] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14355] channel.c: Channel 0x7f0c8008bb10 'SIP/zvonobot-0000009f' allocated [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:15] DEBUG[14355] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:15] DEBUG[14119] chan_sip.c: Hangup call SIP/zvonobot-00000075, SIP callid 564726e17074235c1af6801638e43e42@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14119] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:15] DEBUG[14119] res_rtp_asterisk.c: (0x7f0c9c0840e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14119] res_rtp_asterisk.c: (0x7f0c9c0840e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:15] DEBUG[14119] channel.c: Channel 0x7f0c9c08d3f0 'SIP/zvonobot-00000075' destroying [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Finding handler for channels [Aug 18 10:34:15] DEBUG[14355] res_stasis.c: calls_0: Subscribing to 213121 [Aug 18 10:34:15] DEBUG[14355] stasis/app.c: Channel '213121' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14347] res_stasis.c: calls_0: Subscribing to 213120 [Aug 18 10:34:15] DEBUG[14347] stasis/app.c: Channel '213120' is 1 interested in calls_0 [Aug 18 10:34:15] DEBUG[14485] chan_sip.c: Outgoing Call for 79821116920 [Aug 18 10:34:15] DEBUG[14355] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14347] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:15] DEBUG[14355] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Got SDP version 1145089429 and unique parts [root 1145089429 IN IP4 178.62.121.41] [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Finding handler for 213081 [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking channels create: Didn't match 213081 [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:15] DEBUG[14482] res_ari.c: Checking channels externalMedia: Didn't match 213081 [Aug 18 10:34:15] DEBUG[14482] res_ari.c: No explicit handler found for 213081. Using wildcard channelId. [Aug 18 10:34:15] DEBUG[14485] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14347] http.c: HTTP closing session. Top level [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Outgoing Call for 79821116919 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Outgoing Call for 79821116917 [Aug 18 10:34:15] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP got report of 76 bytes from 178.62.121.41:14927 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213077': is 0 interested in calls_0 [Aug 18 10:34:15] DEBUG[20620] stasis/app.c: channel '213077' unsubscribed from calls_0 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14484] chan_sip.c: Audio is at 15044 [Aug 18 10:34:15] VERBOSE[14484] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] VERBOSE[14484] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] VERBOSE[14484] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Initializing initreq for method INVITE - callid 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116919@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 3 [ 52]: From: ;tag=as69045989 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 6 [ 60]: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:15] DEBUG[14119] stasis.c: Destroying topic. name: cache:288/channel:213077, detail: [Aug 18 10:34:15] DEBUG[14119] stasis.c: Topic 'cache:288/channel:213077': 0x7f0c9c08fc60 destroyed [Aug 18 10:34:15] DEBUG[14119] stasis.c: Destroying topic. name: channel:213077, detail: [Aug 18 10:34:15] DEBUG[14119] stasis.c: Topic 'channel:213077': 0x7f0c9c08f170 destroyed [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1145089429 1145089429 IN IP4 178.62.121.41... OK. [Aug 18 10:34:15] DEBUG[14486] http.c: HTTP opening session. Top level [Aug 18 10:34:15] DEBUG[14486] http.c: HTTP Request URI is /ari/channels/213077 [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:15] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:15] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:15] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:15] DEBUG[14143] channel.c: Channel 0x7f0c78095bd0 'Recorder/ARI-00000029;2' hanging up. Refs: 2 [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282855.378, detail: [Aug 18 10:34:15] VERBOSE[14484] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.378': 0x7f0c300b3540 created [Aug 18 10:34:15] DEBUG[20545] stasis.c: Creating topic. name: cache:434/channel:1629282855.378, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:434/channel:1629282855.378': 0x7f0c3011cb70 created [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: cache:434/channel:1629282855.378, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'cache:434/channel:1629282855.378': 0x7f0c3011cb70 destroyed [Aug 18 10:34:15] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282855.378, detail: [Aug 18 10:34:15] DEBUG[20545] stasis.c: Topic 'channel:1629282855.378': 0x7f0c300b3540 destroyed [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14483] chan_sip.c: Audio is at 12644 [Aug 18 10:34:15] VERBOSE[14483] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:15] VERBOSE[14483] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:15] VERBOSE[14483] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Initializing initreq for method INVITE - callid 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116917@178.62.121.41 SIP/2.0 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 3 [ 52]: From: ;tag=as2489799b [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 6 [ 60]: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:15] VERBOSE[14483] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #87 [Aug 18 10:34:15] DEBUG[14483] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] VERBOSE[14483] dial.c: Called zvonobot/79821116917 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #57 [Aug 18 10:34:15] DEBUG[14484] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:15] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000075', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213077', '')] [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[14486] http.c: match request [ari/channels/213077] with handler [httpstatus] len 10 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:15] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:15] VERBOSE[14484] dial.c: Called zvonobot/79821116919 [Aug 18 10:34:15] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:15] DEBUG[14485] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:15] DEBUG[14485] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:15] VERBOSE[14485] chan_sip.c: Audio is at 14508 [Aug 18 10:34:15] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:16] VERBOSE[14485] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:16] VERBOSE[14485] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:16] VERBOSE[14485] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Initializing initreq for method INVITE - callid 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116920@178.62.121.41 SIP/2.0 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 3 [ 52]: From: ;tag=as3056f2e0 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 6 [ 60]: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:15 GMT [Aug 18 10:34:16] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:15] DEBUG[14486] http.c: match request [ari/channels/213077] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:16] DEBUG[14486] http.c: match request [ari/channels/213077] with handler [ari] len 3 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:16] VERBOSE[14485] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #106 [Aug 18 10:34:16] DEBUG[14485] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a1440) ICE set role failed; no ice instance [Aug 18 10:34:16] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP setting address on RTP instance [Aug 18 10:34:16] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c980a2f60 -- Strict RTP learning after remote address set to: 178.62.121.41:12380 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12380 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb000c478) from 0x7f0c147e2330 to 0x7f0c980a1618 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0109968) from 0x7f0c147e2330 to 0x7f0c980a1618 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb01099f8) from 0x7f0c147e2330 to 0x7f0c980a1618 [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP ignoring duplicate property [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:16] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000049 setting read format path: alaw -> alaw [Aug 18 10:34:16] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000049 setting write format path: alaw -> alaw [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a1440) DTLS - ast_rtp_activate rtp=0x7f0c980a2f60 - setup and perform DTLS' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a2f60) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c980a2f60) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:16] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Strict routing enforced for session 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[12948] res_rtp_asterisk.c: (0x7f0c90008240) RTP 0x7f0c9000c310 -- Received packet from 178.62.121.41:12818, dropping due to strict RTP protection. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117002@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK016b6865 Max-Forwards: 70 From: ;tag=as19ef62c2 To: ;tag=as6089b55c Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[13138] res_rtp_asterisk.c: (0x2c14110) RTP 0x2c17ba0 -- Received packet from 178.62.121.41:15860, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[14486] http.c: Match made with [ari] [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6617ms with no response [Aug 18 10:34:16] VERBOSE[13552] dial.c: SIP/zvonobot-00000049 answered [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Hanging up call 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14493] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[13549] res_rtp_asterisk.c: (0x7f0c94050050) RTP 0x7f0c94051b70 -- Received packet from 178.62.121.41:13802, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Finding handler for channels/213077 [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 343394319 343394319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18958 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6472ms with no response [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Hanging up call 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14493] http.c: HTTP Request URI is /ari/channels/213134?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116906&callerId=74950493843 [Aug 18 10:34:16] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14204] channel.c: Channel 0x2c47a50 'SIP/zvonobot-0000007e' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[14205] channel.c: Channel 0x7f0c10131cc0 'SIP/zvonobot-00000080' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14493] http.c: match request [ari/channels/213134] with handler [httpstatus] len 10 [Aug 18 10:34:16] VERBOSE[13552] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000049 [Aug 18 10:34:16] DEBUG[13552] stasis/app.c: Channel '213038' is 2 interested in calls_0 [Aug 18 10:34:16] VERBOSE[13552] res_rtp_asterisk.c: 0x7f0c980a2f60 -- Strict RTP switching to RTP target address 178.62.121.41:12380 as source [Aug 18 10:34:16] DEBUG[13552] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:16] DEBUG[13552] channel.c: Channel SIP/zvonobot-00000049 setting read format path: ulaw -> alaw [Aug 18 10:34:16] DEBUG[13552] channel.c: Channel SIP/zvonobot-00000049 setting write format path: alaw -> ulaw [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (1) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[14493] http.c: match request [ari/channels/213134] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14493] http.c: match request [ari/channels/213134] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Finding handler for 213077 [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6145ms with no response [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14493] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking channels create: Didn't match 213077 [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14486] res_ari.c: Checking channels externalMedia: Didn't match 213077 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Hanging up call 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Session timer started: 26 - 1591cc604083d1a612552226202481e2@159.65.48.104:5060 1768000ms [Aug 18 10:34:16] DEBUG[14486] res_ari.c: No explicit handler found for 213077. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14212] channel.c: Channel 0x7f0cb008f170 'SIP/zvonobot-00000081' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000007e - start 1629282849.346805 answer 0.000000 end 1629282856.035790 dur 6.688 bill 1629282856.035 dispo NO ANSWER [Aug 18 10:34:16] DEBUG[14281] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:16] DEBUG[14281] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14493] http.c: HTTP consuming request body [Aug 18 10:34:16] VERBOSE[14485] dial.c: Called zvonobot/79821116920 [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Finding handler for channels/213134 [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000080 - start 1629282849.439646 answer 0.000000 end 1629282856.039289 dur 6.599 bill 1629282856.039 dispo NO ANSWER [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000081 - start 1629282849.602133 answer 0.000000 end 1629282856.067591 dur 6.465 bill 1629282856.067 dispo NO ANSWER [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Finding handler for 213134 [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking channels create: Didn't match 213134 [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14493] res_ari.c: Checking channels externalMedia: Didn't match 213134 [Aug 18 10:34:16] DEBUG[14493] res_ari.c: No explicit handler found for 213134. Using wildcard channelId. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7;received=159.65.48.104 From: ;tag=as2618d3a5 To: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a98cb7;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2618d3a5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 (Checking To) --From tag as2618d3a5 --To-tag [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 68f9e13c454d23177446ff49417e37e8@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #91 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #91)) [Aug 18 10:34:16] DEBUG[13695] audiohook.c: Audiohook 0x7f0cb011ab60 has stale audio in its factories. Flushing them both [Aug 18 10:34:16] DEBUG[14497] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14497] http.c: HTTP Request URI is /ari/channels/213137?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116903&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14497] http.c: match request [ari/channels/213137] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14497] http.c: match request [ari/channels/213137] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14497] http.c: match request [ari/channels/213137] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14497] http.c: Match made with [ari] [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116936@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245 Max-Forwards: 70 From: ;tag=as3ee6d51f To: Contact: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 897496002 897496002 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15928 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (1) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 894741588 894741588 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19144 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (1) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116925@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 Max-Forwards: 70 From: ;tag=as0b888cfb To: Contact: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 211868227 211868227 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '77ec81a43645f30730cc74c217742e98@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac077690) DTLS stop [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac077690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac077690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac077690) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cac077690' [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 564726e17074235c1af6801638e43e42@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '564726e17074235c1af6801638e43e42@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c0840e0) DTLS stop [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c0840e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c0840e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c0840e0) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9c0840e0' [Aug 18 10:34:16] DEBUG[14497] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14498] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14498] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:16] DEBUG[14498] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14498] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14498] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14498] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14495] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14495] http.c: HTTP Request URI is /ari/channels/213136?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116904&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14495] http.c: match request [ari/channels/213136] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Finding handler for channels/213137 [Aug 18 10:34:16] DEBUG[14506] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14506] http.c: HTTP Request URI is /ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:16] DEBUG[14506] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14506] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14506] http.c: match request [ari/bridges/45640e14-e267-477d-81ea-fbac374f9677/play] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14506] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Finding handler for 213137 [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking channels create: Didn't match 213137 [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14498] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14498] stasis.c: Creating topic. name: bridge:26acc09b-99c1-4bbb-afbd-344c8a9a505d, detail: [Aug 18 10:34:16] DEBUG[14498] stasis.c: Topic 'bridge:26acc09b-99c1-4bbb-afbd-344c8a9a505d': 0x7f0c10071720 created [Aug 18 10:34:16] DEBUG[14498] stasis.c: Creating topic. name: cache:435/bridge:26acc09b-99c1-4bbb-afbd-344c8a9a505d, detail: [Aug 18 10:34:16] DEBUG[14498] stasis.c: Topic 'cache:435/bridge:26acc09b-99c1-4bbb-afbd-344c8a9a505d': 0x7f0c10128b20 created [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594;received=159.65.48.104 From: ;tag=as6d27c109 To: ;tag=as276fddfb Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63acfefa" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6d27c109 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as276fddfb [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[14497] res_ari.c: Checking channels externalMedia: Didn't match 213137 [Aug 18 10:34:16] DEBUG[13076] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[14495] http.c: match request [ari/channels/213136] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63acfefa" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 (Checking To) --From tag as6d27c109 --To-tag as276fddfb [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (1) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: Allocating new SIP dialog for 18935dc426ec1ef34b5f7bd00e3ddf53@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14493] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac0a2c80' [Aug 18 10:34:16] DEBUG[14495] http.c: match request [ari/channels/213136] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14449] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 1072, ms is 87 [Aug 18 10:34:16] DEBUG[14497] res_ari.c: No explicit handler found for 213137. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14502] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14502] http.c: HTTP Request URI is /ari/channels/213140?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116900&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14502] http.c: match request [ari/channels/213140] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14495] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14510] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1498731843 1498731843 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 18334 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK134ff885;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as31e40966 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20bcc5bd [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1498731843 1498731843 IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 18334 RTP/AVP 0 8 101 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking To) --From tag as31e40966 --To-tag as20bcc5bd [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Stopping retransmission on '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Strict routing enforced for session 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117044@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e151fdb Max-Forwards: 70 From: ;tag=as31e40966 To: ;tag=as20bcc5bd Contact: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) RTP allocated port 10106 [Aug 18 10:34:16] DEBUG[14495] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 From: ;tag=as404b2233 To: ;tag=as0706ba37 Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d4f8c4e;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as404b2233 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0706ba37 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 (Checking To) --From tag as404b2233 --To-tag as0706ba37 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4e0ed8226730ebda5daf0586608708c0@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 From: ;tag=as3d872a68 To: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:16] DEBUG[14498] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[14498] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14498] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14498] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:16] DEBUG[14498] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] DEBUG[14498] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: calling simple_bridge technology constructor [Aug 18 10:34:16] DEBUG[14498] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: calling simple_bridge technology start [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 From: ;tag=as3d872a68 To: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 From: ;tag=as697b28a1 To: ;tag=as7ad66467 Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as697b28a1 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ad66467 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14502] http.c: match request [ari/channels/213140] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Finding handler for channels/213136 [Aug 18 10:34:16] DEBUG[14510] http.c: HTTP Request URI is /ari/channels/213135?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116905&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14512] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[14511] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Finding handler for 213136 [Aug 18 10:34:16] DEBUG[14512] http.c: HTTP Request URI is /ari/channels/213139?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116901&callerId=74950493843 [Aug 18 10:34:16] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking channels create: Didn't match 213136 [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE creating session 0.0.0.0:10106 (10106) [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14502] http.c: match request [ari/channels/213140] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14512] http.c: match request [ari/channels/213139] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14495] res_ari.c: Checking channels externalMedia: Didn't match 213136 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 (Checking To) --From tag as697b28a1 --To-tag as7ad66467 [Aug 18 10:34:16] DEBUG[14511] http.c: HTTP Request URI is /ari/channels/213141?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116899&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14495] res_ari.c: No explicit handler found for 213136. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14512] http.c: match request [ari/channels/213139] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157009242 157009242 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14160 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Finding handler for bridges/45640e14-e267-477d-81ea-fbac374f9677/play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14502] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14512] http.c: match request [ari/channels/213139] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14502] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Finding handler for channels/213140 [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Finding handler for 213140 [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking channels create: Didn't match 213140 [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14502] res_ari.c: Checking channels externalMedia: Didn't match 213140 [Aug 18 10:34:16] DEBUG[14502] res_ari.c: No explicit handler found for 213140. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Finding handler for 45640e14-e267-477d-81ea-fbac374f9677 [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14506] res_ari.c: No explicit handler found for 45640e14-e267-477d-81ea-fbac374f9677. Using wildcard bridgeId. [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Finding handler for play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:16] DEBUG[14506] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:16] DEBUG[14510] http.c: match request [ari/channels/213135] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14512] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14510] http.c: match request [ari/channels/213135] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14510] http.c: match request [ari/channels/213135] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14510] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14510] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Finding handler for channels/213135 [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Finding handler for 213135 [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking channels create: Didn't match 213135 [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14510] res_ari.c: Checking channels externalMedia: Didn't match 213135 [Aug 18 10:34:16] DEBUG[14510] res_ari.c: No explicit handler found for 213135. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14511] http.c: match request [ari/channels/213141] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14498] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:16] DEBUG[14513] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE create [Aug 18 10:34:16] DEBUG[14513] http.c: HTTP Request URI is /ari/channels/213138?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116902&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14513] http.c: match request [ari/channels/213138] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14513] http.c: match request [ari/channels/213138] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14513] http.c: match request [ari/channels/213138] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14513] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14511] http.c: match request [ari/channels/213141] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14511] http.c: match request [ari/channels/213141] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14511] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14511] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14516] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14512] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Finding handler for channels/213141 [Aug 18 10:34:16] DEBUG[14498] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14516] http.c: HTTP Request URI is /ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/addChannel?channel=213038 [Aug 18 10:34:16] DEBUG[14513] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP got report of 76 bytes from 178.62.121.41:10695 [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Finding handler for 213141 [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking channels create: Didn't match 213141 [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14511] res_ari.c: Checking channels externalMedia: Didn't match 213141 [Aug 18 10:34:16] DEBUG[14511] res_ari.c: No explicit handler found for 213141. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6;received=159.65.48.104 From: ;tag=as42dc6c45 To: ;tag=as149154b5 Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc6c45 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as149154b5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 (Checking To) --From tag as42dc6c45 --To-tag as149154b5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Stopping retransmission on '36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117065@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7dc65bb6 Max-Forwards: 70 From: ;tag=as42dc6c45 To: ;tag=as149154b5 Contact: Call-ID: 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Finding handler for channels/213139 [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE add system candidates [Aug 18 10:34:16] DEBUG[14493] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14493] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE add candidate: 159.65.48.104:10106, 2130706431 [Aug 18 10:34:16] DEBUG[14493] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14493] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE add candidate: 10.131.0.10:10106, 2130706431 [Aug 18 10:34:16] DEBUG[14493] rtp_engine.c: RTP instance '0x7f0cac0a2c80' is setup and ready to go [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE stopped [Aug 18 10:34:16] DEBUG[14493] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14493] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14493] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14493] res_rtp_asterisk.c: (0x7f0cac0a2c80) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14493] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14493] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14493] chan_sip.c: SIP call-id changed from '18935dc426ec1ef34b5f7bd00e3ddf53@127.0.1.1:5060' to '1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14493] stasis.c: Creating topic. name: channel:213134, detail: [Aug 18 10:34:16] DEBUG[14493] stasis.c: Topic 'channel:213134': 0x7f0cac047f60 created [Aug 18 10:34:16] DEBUG[14493] stasis.c: Creating topic. name: cache:436/channel:213134, detail: [Aug 18 10:34:16] DEBUG[14493] stasis.c: Topic 'cache:436/channel:213134': 0x7f0cac03ffe0 created [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14517] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[12937] dial.c: SIP/zvonobot-0000000b is busy [Aug 18 10:34:16] DEBUG[14516] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #37 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000000b - start 1629282824.037263 answer 0.000000 end 1629282856.285937 dur 32.248 bill 1629282856.285 dispo BUSY [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #37)) [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116933@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40a1bde8 Max-Forwards: 70 From: ;tag=as52ec131c To: Contact: Call-ID: 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1194373222 1194373222 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[12937] channel.c: Channel 0x7f0c80011850 'SIP/zvonobot-0000000b' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14516] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14516] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/addChannel] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14516] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Finding handler for bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/addChannel [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Finding handler for 26acc09b-99c1-4bbb-afbd-344c8a9a505d [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14516] res_ari.c: No explicit handler found for 26acc09b-99c1-4bbb-afbd-344c8a9a505d. Using wildcard bridgeId. [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Finding handler for addChannel [Aug 18 10:34:16] DEBUG[14516] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:16] DEBUG[14516] stasis/control.c: 213038: Sending channel add_to_bridge command [Aug 18 10:34:16] DEBUG[14517] http.c: HTTP Request URI is /ari/channels/213143?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116897&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14514] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Finding handler for channels/213138 [Aug 18 10:34:16] DEBUG[14286] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[14286] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:16] DEBUG[14517] http.c: match request [ari/channels/213143] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 343394319 343394319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18958 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[14449] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] http.c: match request [ari/channels/213143] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: Allocating new SIP dialog for 5adca20c6313349d0eab7a375f8d696b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14514] http.c: HTTP Request URI is /ari/channels/213142?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116898&callerId=74950493843 [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[13552] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000049 [Aug 18 10:34:16] DEBUG[14517] http.c: match request [ari/channels/213143] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14514] http.c: match request [ari/channels/213142] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14497] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c83a90' [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 From: ;tag=as3a3fa466 To: ;tag=as28515388 Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a3fa466 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as28515388 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 (Checking To) --From tag as3a3fa466 --To-tag as28515388 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) RTP allocated port 13978 [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE creating session 0.0.0.0:13978 (13978) [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE create [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE add system candidates [Aug 18 10:34:16] DEBUG[14497] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14497] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE add candidate: 159.65.48.104:13978, 2130706431 [Aug 18 10:34:16] DEBUG[14514] http.c: match request [ari/channels/213142] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Finding handler for 213139 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #21 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking channels create: Didn't match 213139 [Aug 18 10:34:16] DEBUG[13552] stasis/control.c: 213038: Adding to bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d [Aug 18 10:34:16] DEBUG[13552] stasis/app.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' is 1 interested in calls_0 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #21)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116929@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30e8e9a7 Max-Forwards: 70 From: ;tag=as48744f9a To: Contact: Call-ID: 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 438535394 438535394 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19502 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (2) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1070485752 1070485752 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15860 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5f99fd9e [Aug 18 10:34:16] DEBUG[14497] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14497] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE add candidate: 10.131.0.10:13978, 2130706431 [Aug 18 10:34:16] DEBUG[14497] rtp_engine.c: RTP instance '0x2c83a90' is setup and ready to go [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) ICE stopped [Aug 18 10:34:16] DEBUG[14497] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14497] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14497] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14497] res_rtp_asterisk.c: (0x2c83a90) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14497] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14497] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14497] chan_sip.c: SIP call-id changed from '5adca20c6313349d0eab7a375f8d696b@127.0.1.1:5060' to '5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: Allocating new SIP dialog for 26287fdf7ef5bc5b102a92a950d84cd4@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14511] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c24031c50' [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) RTP allocated port 17578 [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE creating session 0.0.0.0:17578 (17578) [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE create [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE add system candidates [Aug 18 10:34:16] DEBUG[14519] bridge_channel.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: 0x7f0c3c133420(SIP/zvonobot-00000049) is joining [Aug 18 10:34:16] DEBUG[14497] stasis.c: Creating topic. name: channel:213137, detail: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[14517] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14497] stasis.c: Topic 'channel:213137': 0x2c97140 created [Aug 18 10:34:16] DEBUG[14497] stasis.c: Creating topic. name: cache:437/channel:213137, detail: [Aug 18 10:34:16] DEBUG[14497] stasis.c: Topic 'cache:437/channel:213137': 0x2c97b70 created [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1070485752 1070485752 IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15860 RTP/AVP 0 8 101 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 (Checking To) --From tag as4ca0b4bd --To-tag as5f99fd9e [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Stopping retransmission on '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Got SDP version 1070485752 and unique parts [root 1070485752 IN IP4 178.62.121.41] [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1070485752 1070485752 IN IP4 178.62.121.41... OK. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c14110) ICE set role failed; no ice instance [Aug 18 10:34:16] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c14110) RTCP setting address on RTP instance [Aug 18 10:34:16] VERBOSE[20585] res_rtp_asterisk.c: 0x2c17ba0 -- Strict RTP learning after remote address set to: 178.62.121.41:15860 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:15860 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb012fed8) from 0x7f0c147e2330 to 0x2c142e8 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb007f868) from 0x7f0c147e2330 to 0x2c142e8 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0143b38) from 0x7f0c147e2330 to 0x2c142e8 [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c14110) RTCP ignoring duplicate property [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:16] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: Allocating new SIP dialog for 54c7786a4b8d93c56983dc3b0c612523@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14510] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c180962b0' [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) RTP allocated port 13604 [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE creating session 0.0.0.0:13604 (13604) [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE create [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE add system candidates [Aug 18 10:34:16] DEBUG[14510] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14510] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE add candidate: 159.65.48.104:13604, 2130706431 [Aug 18 10:34:16] DEBUG[14510] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14510] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE add candidate: 10.131.0.10:13604, 2130706431 [Aug 18 10:34:16] DEBUG[14510] rtp_engine.c: RTP instance '0x7f0c180962b0' is setup and ready to go [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) ICE stopped [Aug 18 10:34:16] DEBUG[14510] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14510] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14510] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14510] res_rtp_asterisk.c: (0x7f0c180962b0) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14510] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14512] res_ari.c: Checking channels externalMedia: Didn't match 213139 [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000026 setting read format path: alaw -> alaw [Aug 18 10:34:16] DEBUG[14511] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14511] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE add candidate: 159.65.48.104:17578, 2130706431 [Aug 18 10:34:16] DEBUG[14511] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14511] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE add candidate: 10.131.0.10:17578, 2130706431 [Aug 18 10:34:16] DEBUG[14511] rtp_engine.c: RTP instance '0x7f0c24031c50' is setup and ready to go [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) ICE stopped [Aug 18 10:34:16] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 656, ms is 61 [Aug 18 10:34:16] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14514] http.c: match request [ari/channels/213142] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000026 setting write format path: alaw -> alaw [Aug 18 10:34:16] DEBUG[14511] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Finding handler for 213138 [Aug 18 10:34:16] DEBUG[14517] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c14110) DTLS - ast_rtp_activate rtp=0x2c17ba0 - setup and perform DTLS' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c17ba0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x2c17ba0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:16] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Strict routing enforced for session 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117037@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK270cf808 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14511] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14511] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14511] res_rtp_asterisk.c: (0x7f0c24031c50) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14511] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14511] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14511] chan_sip.c: SIP call-id changed from '26287fdf7ef5bc5b102a92a950d84cd4@127.0.1.1:5060' to '29b2374b344692165b25e9de23da23d4@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14514] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking channels create: Didn't match 213138 [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14513] res_ari.c: Checking channels externalMedia: Didn't match 213138 [Aug 18 10:34:16] DEBUG[14513] res_ari.c: No explicit handler found for 213138. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: Allocating new SIP dialog for 58c09ae82dbc011c0264f3f434b1c62e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14495] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4070230' [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) RTP allocated port 13764 [Aug 18 10:34:16] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 656, ms is 61 [Aug 18 10:34:16] DEBUG[14512] res_ari.c: No explicit handler found for 213139. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE creating session 0.0.0.0:13764 (13764) [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE create [Aug 18 10:34:16] VERBOSE[13138] dial.c: SIP/zvonobot-00000026 answered [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Session timer started: 116 - 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 1768000ms [Aug 18 10:34:16] DEBUG[14514] http.c: HTTP consuming request body [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Finding handler for channels/213142 [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Finding handler for channels/213143 [Aug 18 10:34:16] DEBUG[14511] stasis.c: Creating topic. name: channel:213141, detail: [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE add system candidates [Aug 18 10:34:16] DEBUG[14495] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14495] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE add candidate: 159.65.48.104:13764, 2130706431 [Aug 18 10:34:16] DEBUG[14495] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14495] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE add candidate: 10.131.0.10:13764, 2130706431 [Aug 18 10:34:16] DEBUG[14495] rtp_engine.c: RTP instance '0x7f0cb4070230' is setup and ready to go [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) ICE stopped [Aug 18 10:34:16] DEBUG[14495] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14495] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14511] stasis.c: Topic 'channel:213141': 0x7f0c240ed3e0 created [Aug 18 10:34:16] DEBUG[14511] stasis.c: Creating topic. name: cache:438/channel:213141, detail: [Aug 18 10:34:16] DEBUG[14511] stasis.c: Topic 'cache:438/channel:213141': 0x7f0c24140130 created [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Finding handler for 213142 [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking channels create: Didn't match 213142 [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14514] res_ari.c: Checking channels externalMedia: Didn't match 213142 [Aug 18 10:34:16] DEBUG[14514] res_ari.c: No explicit handler found for 213142. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14294] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40' [Aug 18 10:34:16] DEBUG[14294] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:16] DEBUG[13768] bridge_channel.c: Setting 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) state from:0 to:1 [Aug 18 10:34:16] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14495] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14495] res_rtp_asterisk.c: (0x7f0cb4070230) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14495] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14495] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14495] chan_sip.c: SIP call-id changed from '58c09ae82dbc011c0264f3f434b1c62e@127.0.1.1:5060' to '5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14495] stasis.c: Creating topic. name: channel:213136, detail: [Aug 18 10:34:16] DEBUG[14495] stasis.c: Topic 'channel:213136': 0x7f0cb4091780 created [Aug 18 10:34:16] DEBUG[14495] stasis.c: Creating topic. name: cache:439/channel:213136, detail: [Aug 18 10:34:16] VERBOSE[13138] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000026 [Aug 18 10:34:16] DEBUG[13138] stasis/app.c: Channel '213003' is 2 interested in calls_0 [Aug 18 10:34:16] VERBOSE[13138] res_rtp_asterisk.c: 0x2c17ba0 -- Strict RTP switching to RTP target address 178.62.121.41:15860 as source [Aug 18 10:34:16] DEBUG[13138] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:16] DEBUG[13138] channel.c: Channel SIP/zvonobot-00000026 setting read format path: ulaw -> alaw [Aug 18 10:34:16] DEBUG[13138] channel.c: Channel SIP/zvonobot-00000026 setting write format path: alaw -> ulaw [Aug 18 10:34:16] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13768] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: pulling 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) [Aug 18 10:34:16] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14520] http.c: HTTP opening session. Top level [Aug 18 10:34:16] VERBOSE[13768] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 left 'simple_bridge' stasis-bridge [Aug 18 10:34:16] DEBUG[13768] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) is leaving simple_bridge technology [Aug 18 10:34:16] DEBUG[14294] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14519] bridge_channel.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: pushing 0x7f0c3c133420(SIP/zvonobot-00000049) [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 From: ;tag=as2eb39fa6 To: ;tag=as7b46504f Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2eb39fa6 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7b46504f [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 (Checking To) --From tag as2eb39fa6 --To-tag as7b46504f [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Finding handler for 213143 [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking channels create: Didn't match 213143 [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14517] res_ari.c: Checking channels externalMedia: Didn't match 213143 [Aug 18 10:34:16] DEBUG[14517] res_ari.c: No explicit handler found for 213143. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: Allocating new SIP dialog for 2749b4742eb7658f10095b1d7778bc77@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14502] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c080681f0' [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) RTP allocated port 11616 [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE creating session 0.0.0.0:11616 (11616) [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE create [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE add system candidates [Aug 18 10:34:16] DEBUG[14502] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14502] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE add candidate: 159.65.48.104:11616, 2130706431 [Aug 18 10:34:16] DEBUG[14502] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14502] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE add candidate: 10.131.0.10:11616, 2130706431 [Aug 18 10:34:16] DEBUG[14502] rtp_engine.c: RTP instance '0x7f0c080681f0' is setup and ready to go [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) ICE stopped [Aug 18 10:34:16] DEBUG[14502] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14502] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14502] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14502] res_rtp_asterisk.c: (0x7f0c080681f0) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14502] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:16] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:16] VERBOSE[14519] bridge_channel.c: Channel SIP/zvonobot-00000049 joined 'simple_bridge' stasis-bridge <26acc09b-99c1-4bbb-afbd-344c8a9a505d> [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14502] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14520] http.c: HTTP Request URI is /ari/channels/213008 [Aug 18 10:34:16] DEBUG[14520] http.c: match request [ari/channels/213008] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14520] http.c: match request [ari/channels/213008] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14520] http.c: match request [ari/channels/213008] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14520] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Finding handler for channels/213008 [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Finding handler for 213008 [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking channels create: Didn't match 213008 [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14520] res_ari.c: Checking channels externalMedia: Didn't match 213008 [Aug 18 10:34:16] DEBUG[14520] res_ari.c: No explicit handler found for 213008. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14521] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14495] stasis.c: Topic 'cache:439/channel:213136': 0x7f0cb40921c0 created [Aug 18 10:34:16] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 From: ;tag=as19ef62c2 To: ;tag=as6089b55c Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1145089429 1145089429 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12380 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:16] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37955652;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14521] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:16] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as19ef62c2 [Aug 18 10:34:16] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 720, ms is 65 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6089b55c [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1145089429 1145089429 IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12380 RTP/AVP 0 8 101 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for Snoop/213008-0000000a - start 1629282839.754320 answer 1629282839.754320 end 1629282856.495426 dur 16.741 bill 16.741 dispo ANSWERED [Aug 18 10:34:16] DEBUG[14521] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[13768] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[14521] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14510] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14510] chan_sip.c: SIP call-id changed from '54c7786a4b8d93c56983dc3b0c612523@127.0.1.1:5060' to '7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14521] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:16] DEBUG[13768] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 (Checking To) --From tag as19ef62c2 --To-tag as6089b55c [Aug 18 10:34:16] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13768] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[13227] res_rtp_asterisk.c: (0x7f0c28011240) RTP audio difference is 1040, ms is 85 [Aug 18 10:34:16] DEBUG[14521] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[13768] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13768] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14502] chan_sip.c: SIP call-id changed from '2749b4742eb7658f10095b1d7778bc77@127.0.1.1:5060' to '2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14502] stasis.c: Creating topic. name: channel:213140, detail: [Aug 18 10:34:16] DEBUG[14502] stasis.c: Topic 'channel:213140': 0x7f0c08041100 created [Aug 18 10:34:16] DEBUG[14502] stasis.c: Creating topic. name: cache:440/channel:213140, detail: [Aug 18 10:34:16] DEBUG[14502] stasis.c: Topic 'cache:440/channel:213140': 0x7f0c080772b0 created [Aug 18 10:34:16] DEBUG[14420] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14510] stasis.c: Creating topic. name: channel:213135, detail: [Aug 18 10:34:16] DEBUG[13768] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a is already using the new technology. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Stopping retransmission on '1591cc604083d1a612552226202481e2@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Strict routing enforced for session 1591cc604083d1a612552226202481e2@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:16] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:16] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117002@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0ae8e4bb Max-Forwards: 70 From: ;tag=as19ef62c2 To: ;tag=as6089b55c Contact: Call-ID: 1591cc604083d1a612552226202481e2@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14293] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:16] DEBUG[14298] channel.c: Channel 0x7f0c400b0ff0 'Recorder/ARI-00000033;2' allocated [Aug 18 10:34:16] DEBUG[14298] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14519] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[14293] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14091] channel.c: Channel 0x7f0c180d6770 'Announcer/ARI-00000032;2' allocated [Aug 18 10:34:16] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14521] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14521] stasis.c: Creating topic. name: bridge:cd02ec13-331f-440c-a360-837dbdfdba5e, detail: [Aug 18 10:34:16] DEBUG[14521] stasis.c: Topic 'bridge:cd02ec13-331f-440c-a360-837dbdfdba5e': 0x7f0c400575c0 created [Aug 18 10:34:16] DEBUG[14521] stasis.c: Creating topic. name: cache:441/bridge:cd02ec13-331f-440c-a360-837dbdfdba5e, detail: [Aug 18 10:34:16] DEBUG[14521] stasis.c: Topic 'cache:441/bridge:cd02ec13-331f-440c-a360-837dbdfdba5e': 0x7f0c4008b510 created [Aug 18 10:34:16] DEBUG[14521] bridge_native_rtp.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[14519] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14091] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:16] DEBUG[14091] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000032;1' [Aug 18 10:34:16] DEBUG[14523] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c180c90e0(Announcer/ARI-00000032;2) is joining [Aug 18 10:34:16] DEBUG[14519] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14522] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c4007ef90(Recorder/ARI-00000033;2) is joining [Aug 18 10:34:16] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14519] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14521] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14510] stasis.c: Topic 'channel:213135': 0x7f0c18096a40 created [Aug 18 10:34:16] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14519] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] DEBUG[14519] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d is already using the new technology. [Aug 18 10:34:16] DEBUG[14510] stasis.c: Creating topic. name: cache:442/channel:213135, detail: [Aug 18 10:34:16] DEBUG[14510] stasis.c: Topic 'cache:442/channel:213135': 0x7f0c180c8af0 created [Aug 18 10:34:16] DEBUG[13768] bridge_channel.c: Bridge is returning 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) to write format slin16 [Aug 18 10:34:16] DEBUG[14519] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: 0x7f0c3c133420(SIP/zvonobot-00000049) is joining simple_bridge technology [Aug 18 10:34:16] DEBUG[14521] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:16] DEBUG[13208] audiohook.c: Audiohook 0x7f0c2402b640 has stale audio in its factories. Flushing them both [Aug 18 10:34:16] DEBUG[13768] channel.c: Channel UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40 setting write format path: slin16 -> slin16 [Aug 18 10:34:16] DEBUG[14522] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: pushing 0x7f0c4007ef90(Recorder/ARI-00000033;2) [Aug 18 10:34:16] DEBUG[13768] stasis/control.c: robot_213008, e0573cd4-75f6-4425-a1e4-83029f01aa9a: Channel was departed from bridge [Aug 18 10:34:16] DEBUG[13768] stasis/app.c: bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a': is 3 interested in calls_0 [Aug 18 10:34:16] DEBUG[14519] res_rtp_asterisk.c: (0x7f0c980a1440) RTP changing ssrc from 1769035169 to 457434242 due to a source change [Aug 18 10:34:16] DEBUG[13552] stasis/app.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' is 2 interested in calls_0 [Aug 18 10:34:16] DEBUG[13739] stasis/control.c: robot_213008: Channel departing bridge [Aug 18 10:34:16] DEBUG[13768] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:16] DEBUG[13739] bridge.c: Waiting for 0x7f0c74033a40(UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40) bridge thread to die. [Aug 18 10:34:16] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14516] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:16] DEBUG[14522] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:16] VERBOSE[14522] bridge_channel.c: Channel Recorder/ARI-00000033;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:16] DEBUG[14516] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14521] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14523] bridge_channel.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: pushing 0x7f0c180c90e0(Announcer/ARI-00000032;2) [Aug 18 10:34:16] DEBUG[13739] stasis/app.c: channel 'robot_213008': is 1 interested in calls_0 [Aug 18 10:34:16] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP got report of 100 bytes from 178.62.121.41:14675 [Aug 18 10:34:16] DEBUG[14524] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[13739] channel.c: Channel 0x7f0ca806b9f0 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40' hanging up. Refs: 2 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 From: ;tag=as04f0121c To: ;tag=as31dfa2de Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as04f0121c [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as31dfa2de [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14523] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[14524] http.c: HTTP Request URI is /ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/record?name=213038_BKsPPDtGadkoThbYiyRxVjYHUBQlBmjs&format=wav [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 (Checking To) --From tag as04f0121c --To-tag as31dfa2de [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Stopping retransmission on '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117036@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK649b50ae Max-Forwards: 70 From: ;tag=as04f0121c To: ;tag=as31dfa2de Contact: Call-ID: 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #61 (5) INVITE - 5 [Aug 18 10:34:16] VERBOSE[14523] bridge_channel.c: Channel Announcer/ARI-00000032;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #61)) [Aug 18 10:34:16] DEBUG[14521] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116928@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9 Max-Forwards: 70 From: ;tag=as03ee25b2 To: Contact: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1861002524 1861002524 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15988 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14521] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: calling simple_bridge technology constructor [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14524] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/record] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1566674665 1566674665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11680 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #60 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #60)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116927@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK38f9f1b3 Max-Forwards: 70 From: ;tag=as2c4993ac To: Contact: Call-ID: 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2022961929 2022961929 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12990 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: Allocating new SIP dialog for 163c792019e533bf0d29ee5769e7234c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14513] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c0a9e10' [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) RTP allocated port 14208 [Aug 18 10:34:16] DEBUG[14521] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: calling simple_bridge technology start [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE creating session 0.0.0.0:14208 (14208) [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE create [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add system candidates [Aug 18 10:34:16] DEBUG[14513] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14513] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add candidate: 159.65.48.104:14208, 2130706431 [Aug 18 10:34:16] DEBUG[14513] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14513] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE add candidate: 10.131.0.10:14208, 2130706431 [Aug 18 10:34:16] DEBUG[14513] rtp_engine.c: RTP instance '0x7f0c2c0a9e10' is setup and ready to go [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE stopped [Aug 18 10:34:16] DEBUG[14513] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14513] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14513] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14513] res_rtp_asterisk.c: (0x7f0c2c0a9e10) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14513] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[13059] app.c: One waitfor failed, trying another [Aug 18 10:34:16] DEBUG[14522] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d'. Checking compatability for channels 'SIP/zvonobot-00000006' and 'Recorder/ARI-00000033;2' [Aug 18 10:34:16] DEBUG[14522] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as could not get details [Aug 18 10:34:16] DEBUG[14522] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14522] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8;received=159.65.48.104 From: ;tag=as2f5156ef To: ;tag=as4a4114fd Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dd12f75" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[14525] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14522] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2f5156ef [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4a4114fd [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dd12f75" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 (Checking To) --From tag as2f5156ef --To-tag as4a4114fd [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6097ms with no response [Aug 18 10:34:16] WARNING[20585] chan_sip.c: Hanging up call 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14525] http.c: HTTP Request URI is /ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/addChannel?channel=213003 [Aug 18 10:34:16] DEBUG[14158] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14524] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/record] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14525] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14262] channel.c: Channel 0x7f0c7c013ef0 'SIP/zvonobot-00000082' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[14522] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000082 - start 1629282850.221146 answer 0.000000 end 1629282856.720788 dur 6.499 bill 1629282856.720 dispo NO ANSWER [Aug 18 10:34:16] DEBUG[14521] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:16] DEBUG[14525] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14524] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/record] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14522] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d is already using the new technology. [Aug 18 10:34:16] DEBUG[14513] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14522] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c4007ef90(Recorder/ARI-00000033;2) is joining simple_bridge technology [Aug 18 10:34:16] DEBUG[14522] channel.c: Channel Recorder/ARI-00000033;2 setting read format path: slin -> slin [Aug 18 10:34:16] DEBUG[14522] channel.c: Channel SIP/zvonobot-00000006 setting write format path: slin -> ulaw [Aug 18 10:34:16] DEBUG[14522] channel.c: Channel SIP/zvonobot-00000006 setting read format path: ulaw -> slin [Aug 18 10:34:16] DEBUG[14522] channel.c: Channel Recorder/ARI-00000033;2 setting write format path: slin -> slin [Aug 18 10:34:16] DEBUG[14525] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/addChannel] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14525] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14524] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14521] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Finding handler for bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/addChannel [Aug 18 10:34:16] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTCP got report of 100 bytes from 178.62.121.41:18793 [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: Allocating new SIP dialog for 3e005f2f0a7c02e778b964cf78e9a63f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] WARNING[14444] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000031;1 [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Finding handler for bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Finding handler for cd02ec13-331f-440c-a360-837dbdfdba5e [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14525] res_ari.c: No explicit handler found for cd02ec13-331f-440c-a360-837dbdfdba5e. Using wildcard bridgeId. [Aug 18 10:34:16] DEBUG[14122] chan_sip.c: Hangup call SIP/zvonobot-00000076, SIP callid 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14122] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:16] DEBUG[14122] res_rtp_asterisk.c: (0x7f0cb408bdf0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[14122] res_rtp_asterisk.c: (0x7f0cb408bdf0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[14122] channel.c: Channel 0x7f0cb4080db0 'SIP/zvonobot-00000076' destroying [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Finding handler for addChannel [Aug 18 10:34:16] DEBUG[13167] chan_sip.c: Hangup call SIP/zvonobot-00000028, SIP callid 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[13167] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[13167] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[13167] channel.c: Channel 0x7f0c30038fd0 'SIP/zvonobot-00000028' destroying [Aug 18 10:34:16] DEBUG[12965] chan_sip.c: Hangup call SIP/zvonobot-00000011, SIP callid 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[12965] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[12965] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[12965] channel.c: Channel 0x7f0cb401fdb0 'SIP/zvonobot-00000011' destroying [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213083': is 0 interested in calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213083' unsubscribed from calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: cache:289/channel:213083, detail: [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'cache:289/channel:213083': 0x7f0cb4083570 destroyed [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: channel:213083, detail: [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'channel:213083': 0x7f0cb4082b30 destroyed [Aug 18 10:34:16] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14397] channel.c: Channel 0x7f0c3c12ca30 'Recorder/ARI-00000035;1' allocated [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14133] channel.c: Channel 0x7f0c1c13de00 'Announcer/ARI-00000028;2' destroying [Aug 18 10:34:16] DEBUG[14141] chan_sip.c: Hangup call SIP/zvonobot-00000077, SIP callid 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14141] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:16] DEBUG[14141] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[14141] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[14141] channel.c: Channel 0x7f0cb0160ed0 'SIP/zvonobot-00000077' destroying [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14397] stasis.c: Creating topic. name: channel:1629282856.385, detail: [Aug 18 10:34:16] DEBUG[14397] stasis.c: Topic 'channel:1629282856.385': 0x7f0c3c11c2f0 created [Aug 18 10:34:16] DEBUG[14397] stasis.c: Creating topic. name: cache:443/channel:1629282856.385, detail: [Aug 18 10:34:16] DEBUG[14397] stasis.c: Topic 'cache:443/channel:1629282856.385': 0x7f0c3c089750 created [Aug 18 10:34:16] DEBUG[14273] channel.c: Channel 0x7f0c90025910 'Announcer/ARI-0000002b;2' destroying [Aug 18 10:34:16] DEBUG[14525] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:16] DEBUG[14304] channel.c: Channel 0x7f0c7804c540 'Recorder/ARI-00000034;1' allocated [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[14304] stasis.c: Creating topic. name: channel:1629282856.386, detail: [Aug 18 10:34:16] DEBUG[14304] stasis.c: Topic 'channel:1629282856.386': 0x7f0c78070280 created [Aug 18 10:34:16] DEBUG[14304] stasis.c: Creating topic. name: cache:444/channel:1629282856.386, detail: [Aug 18 10:34:16] DEBUG[14304] stasis.c: Topic 'cache:444/channel:1629282856.386': 0x7f0c7803d9e0 created [Aug 18 10:34:16] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '212978': is 0 interested in calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '212978' unsubscribed from calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: cache:24/channel:212978, detail: [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'cache:24/channel:212978': 0x7f0cb4023360 destroyed [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213004': is 0 interested in calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213004' unsubscribed from calls_0 [Aug 18 10:34:16] DEBUG[14526] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14200] channel.c: Channel 0x7f0ca0055050 'Announcer/ARI-00000037;1' allocated [Aug 18 10:34:16] DEBUG[14200] stasis.c: Creating topic. name: channel:1629282856.387, detail: [Aug 18 10:34:16] DEBUG[14200] stasis.c: Topic 'channel:1629282856.387': 0x7f0ca0036030 created [Aug 18 10:34:16] DEBUG[14200] stasis.c: Creating topic. name: cache:445/channel:1629282856.387, detail: [Aug 18 10:34:16] DEBUG[14200] stasis.c: Topic 'cache:445/channel:1629282856.387': 0x7f0ca01512f0 created [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14525] stasis/control.c: 213003: Sending channel add_to_bridge command [Aug 18 10:34:16] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14403] channel.c: Channel 0x7f0c40085a70 'Recorder/ARI-00000036;1' allocated [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14432] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14432] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:16] DEBUG[14432] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:16] DEBUG[14432] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:16] DEBUG[14432] channel.c: Channel Announcer/ARI-0000002f;1 setting write format path: slin -> slin [Aug 18 10:34:16] DEBUG[14432] channel.c: Channel 0x7f0c9c09b130 'Announcer/ARI-0000002f;1' hanging up. Refs: 2 [Aug 18 10:34:16] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14514] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280e6850' [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14133] stasis.c: Destroying topic. name: cache:291/channel:1629282845.249, detail: [Aug 18 10:34:16] DEBUG[14273] stasis.c: Destroying topic. name: cache:323/channel:1629282848.276, detail: [Aug 18 10:34:16] DEBUG[14273] stasis.c: Topic 'cache:323/channel:1629282848.276': 0x7f0c900268e0 destroyed [Aug 18 10:34:16] DEBUG[14273] stasis.c: Destroying topic. name: channel:1629282848.276, detail: [Aug 18 10:34:16] DEBUG[14273] stasis.c: Topic 'channel:1629282848.276': 0x7f0c900288b0 destroyed [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Finding handler for 26acc09b-99c1-4bbb-afbd-344c8a9a505d [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14524] res_ari.c: No explicit handler found for 26acc09b-99c1-4bbb-afbd-344c8a9a505d. Using wildcard bridgeId. [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Finding handler for record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:16] DEBUG[14524] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:16] DEBUG[14524] stasis.c: Creating topic. name: channel:1629282856.390, detail: [Aug 18 10:34:16] DEBUG[14524] stasis.c: Topic 'channel:1629282856.390': 0x7f0c7c00d640 created [Aug 18 10:34:16] DEBUG[14524] stasis.c: Creating topic. name: cache:446/channel:1629282856.390, detail: [Aug 18 10:34:16] DEBUG[14524] stasis.c: Topic 'cache:446/channel:1629282856.390': 0x7f0c7c0d09e0 created [Aug 18 10:34:16] DEBUG[14526] http.c: HTTP Request URI is /ari/channels/213083 [Aug 18 10:34:16] DEBUG[14528] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14133] stasis.c: Topic 'cache:291/channel:1629282845.249': 0x7f0c1c140580 destroyed [Aug 18 10:34:16] DEBUG[14528] http.c: HTTP Request URI is /ari/channels/213004 [Aug 18 10:34:16] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282856.388, detail: [Aug 18 10:34:16] DEBUG[14528] http.c: match request [ari/channels/213004] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14403] stasis.c: Creating topic. name: channel:1629282856.389, detail: [Aug 18 10:34:16] DEBUG[14403] stasis.c: Topic 'channel:1629282856.389': 0x7f0c40070450 created [Aug 18 10:34:16] DEBUG[14403] stasis.c: Creating topic. name: cache:447/channel:1629282856.389, detail: [Aug 18 10:34:16] DEBUG[14403] stasis.c: Topic 'cache:447/channel:1629282856.389': 0x7f0c40038fa0 created [Aug 18 10:34:16] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13138] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000026 [Aug 18 10:34:16] DEBUG[14528] http.c: match request [ari/channels/213004] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13138] stasis/control.c: 213003: Adding to bridge cd02ec13-331f-440c-a360-837dbdfdba5e [Aug 18 10:34:16] DEBUG[14528] http.c: match request [ari/channels/213004] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14133] stasis.c: Destroying topic. name: channel:1629282845.249, detail: [Aug 18 10:34:16] DEBUG[14133] stasis.c: Topic 'channel:1629282845.249': 0x7f0c1c13fb50 destroyed [Aug 18 10:34:16] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14526] http.c: match request [ari/channels/213083] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14528] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[13167] stasis.c: Destroying topic. name: cache:59/channel:213004, detail: [Aug 18 10:34:16] DEBUG[13167] stasis.c: Topic 'cache:59/channel:213004': 0x7f0c300ab560 destroyed [Aug 18 10:34:16] DEBUG[13138] stasis/app.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e' is 1 interested in calls_0 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213080': is 0 interested in calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis/app.c: channel '213080' unsubscribed from calls_0 [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: cache:290/channel:213080, detail: [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'cache:290/channel:213080': 0x7f0cb0058100 destroyed [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Finding handler for channels/213004 [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20545] stasis.c: Topic 'channel:1629282856.388': 0x7f0c3002cab0 created [Aug 18 10:34:16] DEBUG[20545] stasis.c: Creating topic. name: cache:448/channel:1629282856.388, detail: [Aug 18 10:34:16] DEBUG[20545] stasis.c: Topic 'cache:448/channel:1629282856.388': 0x7f0c300ab560 created [Aug 18 10:34:16] DEBUG[20545] stasis.c: Destroying topic. name: cache:448/channel:1629282856.388, detail: [Aug 18 10:34:16] DEBUG[20545] stasis.c: Topic 'cache:448/channel:1629282856.388': 0x7f0c300ab560 destroyed [Aug 18 10:34:16] DEBUG[14527] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14526] http.c: match request [ari/channels/213083] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: Allocating new SIP dialog for 6b56812d2b8c706c27a91d9c52e6f75b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14513] chan_sip.c: SIP call-id changed from '163c792019e533bf0d29ee5769e7234c@127.0.1.1:5060' to '30cb159f67289df002568fe9006f4752@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14517] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c30041010' [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) RTP allocated port 18198 [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE creating session 0.0.0.0:18198 (18198) [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE create [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) RTP allocated port 17048 [Aug 18 10:34:16] DEBUG[14527] http.c: HTTP Request URI is /ari/channels/212978 [Aug 18 10:34:16] DEBUG[14523] bridge_native_rtp.c: Bridge 'aba705f1-c39f-408a-8a02-8c7f66ee7c7d' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[14530] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282856.388, detail: [Aug 18 10:34:16] DEBUG[14527] http.c: match request [ari/channels/212978] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14523] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14527] http.c: match request [ari/channels/212978] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14527] http.c: match request [ari/channels/212978] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14527] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14529] bridge_channel.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: 0x7f0c1014bce0(SIP/zvonobot-00000026) is joining [Aug 18 10:34:16] DEBUG[20545] stasis.c: Topic 'channel:1629282856.388': 0x7f0c3002cab0 destroyed [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050;received=159.65.48.104 From: ;tag=as22570a36 To: ;tag=as1a2466aa Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="691c573f" Content-Length: 0 <-------------> [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK346a4050;received=159.65.48.104 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22570a36 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1a2466aa [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:16] DEBUG[14526] http.c: match request [ari/channels/213083] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14298] res_stasis_recording.c: 1629282851.315: Sending record(212969_yHqbSyiGmTqhcwsGdMOSOdmkDrfdyHSx.wav) command [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: channel:212978, detail: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:16] DEBUG[14526] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="691c573f" [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE creating session 0.0.0.0:17048 (17048) [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE create [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE add system candidates [Aug 18 10:34:16] DEBUG[14514] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14514] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE add candidate: 159.65.48.104:17048, 2130706431 [Aug 18 10:34:16] DEBUG[14514] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14514] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE add candidate: 10.131.0.10:17048, 2130706431 [Aug 18 10:34:16] DEBUG[14514] rtp_engine.c: RTP instance '0x7f0c280e6850' is setup and ready to go [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) ICE stopped [Aug 18 10:34:16] DEBUG[14514] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[14514] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[14514] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14514] res_rtp_asterisk.c: (0x7f0c280e6850) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14514] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14514] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14514] chan_sip.c: SIP call-id changed from '3e005f2f0a7c02e778b964cf78e9a63f@127.0.1.1:5060' to '1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14514] stasis.c: Creating topic. name: channel:213142, detail: [Aug 18 10:34:16] DEBUG[14514] stasis.c: Topic 'channel:213142': 0x7f0c280efa80 created [Aug 18 10:34:16] DEBUG[14514] stasis.c: Creating topic. name: cache:449/channel:213142, detail: [Aug 18 10:34:16] DEBUG[14514] stasis.c: Topic 'cache:449/channel:213142': 0x7f0c28104f30 created [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Finding handler for 213004 [Aug 18 10:34:16] DEBUG[14298] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking channels create: Didn't match 213004 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:16] DEBUG[14298] http.c: HTTP closing session. Top level [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE add system candidates [Aug 18 10:34:16] DEBUG[14517] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:16] DEBUG[14517] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE add candidate: 159.65.48.104:18198, 2130706431 [Aug 18 10:34:16] DEBUG[14517] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:16] DEBUG[14517] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE add candidate: 10.131.0.10:18198, 2130706431 [Aug 18 10:34:16] DEBUG[14517] rtp_engine.c: RTP instance '0x7f0c30041010' is setup and ready to go [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) ICE stopped [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Finding handler for channels/213083 [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Finding handler for 213083 [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking channels create: Didn't match 213083 [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14526] res_ari.c: Checking channels externalMedia: Didn't match 213083 [Aug 18 10:34:16] DEBUG[14526] res_ari.c: No explicit handler found for 213083. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14531] app.c: play_and_record: , /var/spool/asterisk/recording/212969_yHqbSyiGmTqhcwsGdMOSOdmkDrfdyHSx, 'wav' [Aug 18 10:34:16] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14513] stasis.c: Creating topic. name: channel:213138, detail: [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'channel:212978': 0x7f0cb40211f0 destroyed [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Finding handler for channels/212978 [Aug 18 10:34:16] DEBUG[14532] http.c: HTTP opening session. Top level [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 (Checking To) --From tag as22570a36 --To-tag as1a2466aa [Aug 18 10:34:16] DEBUG[14530] http.c: HTTP Request URI is /ari/channels/213080 [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14523] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14528] res_ari.c: Checking channels externalMedia: Didn't match 213004 [Aug 18 10:34:16] DEBUG[14528] res_ari.c: No explicit handler found for 213004. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14531] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:16] VERBOSE[14531] app.c: x=0, open writing: /var/spool/asterisk/recording/212969_yHqbSyiGmTqhcwsGdMOSOdmkDrfdyHSx format: wav, 0x7f0c90040870 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 0c08b1570fa732272364833678dc04bb@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14532] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:16] DEBUG[14532] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: channel:213080, detail: [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (3) INVITE - 5 [Aug 18 10:34:16] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000076', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213083', '')] [Aug 18 10:34:16] DEBUG[14517] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'channel:213080': 0x7f0cb0162c50 destroyed [Aug 18 10:34:16] DEBUG[14532] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14523] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[14523] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Finding handler for 212978 [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking channels create: Didn't match 212978 [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14527] res_ari.c: Checking channels externalMedia: Didn't match 212978 [Aug 18 10:34:16] DEBUG[14527] res_ari.c: No explicit handler found for 212978. Using wildcard channelId. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14523] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d is already using the new technology. [Aug 18 10:34:16] DEBUG[14532] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[14523] bridge.c: Bridge aba705f1-c39f-408a-8a02-8c7f66ee7c7d: 0x7f0c180c90e0(Announcer/ARI-00000032;2) is joining simple_bridge technology [Aug 18 10:34:16] DEBUG[14513] stasis.c: Topic 'channel:213138': 0x7f0c2c0f2b40 created [Aug 18 10:34:16] DEBUG[14532] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (5) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:16] DEBUG[14530] http.c: match request [ari/channels/213080] with handler [httpstatus] len 10 [Aug 18 10:34:16] DEBUG[14091] res_stasis_playback.c: 1629282851.313: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1157293889 1157293889 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14091] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:16] DEBUG[14517] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (4) INVITE - 5 [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Finding handler for bridges [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:16] DEBUG[14513] stasis.c: Creating topic. name: cache:450/channel:213138, detail: [Aug 18 10:34:16] DEBUG[14513] stasis.c: Topic 'cache:450/channel:213138': 0x7f0c2c0b34b0 created [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4720589f501ec99f23aa211a55fadfa0@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb408bdf0) DTLS stop [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb408bdf0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb408bdf0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb408bdf0) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:16] DEBUG[14530] http.c: match request [ari/channels/213080] with handler [phoneprov] len 9 [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb408bdf0' [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2ed196de6b5bce76068c31893b4cecbc@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) DTLS stop [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30031690) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:16] DEBUG[14530] http.c: match request [ari/channels/213080] with handler [ari] len 3 [Aug 18 10:34:16] DEBUG[20620] stasis.c: Destroying topic. name: channel:213004, detail: [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c30031690' [Aug 18 10:34:16] DEBUG[14517] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:16] DEBUG[14530] http.c: Match made with [ari] [Aug 18 10:34:16] DEBUG[20620] stasis.c: Topic 'channel:213004': 0x7f0c300ab380 destroyed [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:16] DEBUG[14517] res_rtp_asterisk.c: (0x7f0c30041010) RTCP setup on RTP instance [Aug 18 10:34:16] VERBOSE[14517] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:16] DEBUG[14517] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:16] DEBUG[14517] chan_sip.c: SIP call-id changed from '6b56812d2b8c706c27a91d9c52e6f75b@127.0.1.1:5060' to '5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060' [Aug 18 10:34:16] DEBUG[14517] stasis.c: Creating topic. name: channel:213143, detail: [Aug 18 10:34:16] DEBUG[14517] stasis.c: Topic 'channel:213143': 0x7f0c30030ac0 created [Aug 18 10:34:16] DEBUG[14517] stasis.c: Creating topic. name: cache:451/channel:213143, detail: [Aug 18 10:34:16] DEBUG[14517] stasis.c: Topic 'cache:451/channel:213143': 0x7f0c3010d7f0 created [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060 [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:16] DEBUG[14529] bridge_channel.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: pushing 0x7f0c1014bce0(SIP/zvonobot-00000026) [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Finding handler for channels/213080 [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Finding handler for channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Finding handler for 213080 [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking channels create: Didn't match 213080 [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:16] DEBUG[14530] res_ari.c: Checking channels externalMedia: Didn't match 213080 [Aug 18 10:34:16] DEBUG[14530] res_ari.c: No explicit handler found for 213080. Using wildcard channelId. [Aug 18 10:34:16] DEBUG[14414] channel.c: Channel 0x7f0c8009df10 'SIP/zvonobot-000000a0' allocated [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:16] DEBUG[14414] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5d9ca65b1ec91d545293acc40f191b77@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[14532] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:16] DEBUG[14532] stasis.c: Creating topic. name: bridge:fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1, detail: [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS stop [Aug 18 10:34:16] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[14532] stasis.c: Topic 'bridge:fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1': 0x7f0ca8117e90 created [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[14532] stasis.c: Creating topic. name: cache:452/bridge:fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1, detail: [Aug 18 10:34:16] DEBUG[14532] stasis.c: Topic 'cache:452/bridge:fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1': 0x7f0ca8112690 created [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb401aa90) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:16] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:16] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:16] VERBOSE[14529] bridge_channel.c: Channel SIP/zvonobot-00000026 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb401aa90' [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Destroying SIP dialog 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS stop [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282856.394, detail: [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:16] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE RTP transport deallocating [Aug 18 10:34:16] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb01036b0' [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (4) INVITE - 5 [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:16] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116925@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07a51ed9 Max-Forwards: 70 From: ;tag=as0b888cfb To: Contact: Call-ID: 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 211868227 211868227 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17556 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:16] DEBUG[14532] bridge_native_rtp.c: Bridge 'fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1' can not use native RTP bridge as two channels are required [Aug 18 10:34:16] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:16] DEBUG[14532] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:16] DEBUG[14532] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:16] DEBUG[20545] stasis.c: Topic 'channel:1629282856.394': 0x7f0c300fcf90 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:453/channel:1629282856.394, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:453/channel:1629282856.394': 0x7f0c300fd040 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:453/channel:1629282856.394, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:453/channel:1629282856.394': 0x7f0c300fd040 destroyed [Aug 18 10:34:17] DEBUG[14532] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:17] DEBUG[14414] res_stasis.c: calls_0: Subscribing to 213124 [Aug 18 10:34:17] DEBUG[14414] stasis/app.c: Channel '213124' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14532] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14414] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14532] bridge.c: Bridge fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1: calling simple_bridge technology constructor [Aug 18 10:34:17] DEBUG[14414] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Outgoing Call for 79821116916 [Aug 18 10:34:17] DEBUG[14532] bridge.c: Bridge fabdbb67-15b9-4236-b1ee-ece1b0cc7ba1: calling simple_bridge technology start [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[13141] res_rtp_asterisk.c: (0x7f0c0801b610) RTP 0x7f0c0801f290 -- Received packet from 178.62.121.41:19176, dropping due to strict RTP protection. [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[13035] res_rtp_asterisk.c: (0x7f0c40018bf0) RTP 0x7f0c4001c8f0 -- Received packet from 178.62.121.41:16296, dropping due to strict RTP protection. [Aug 18 10:34:17] DEBUG[14433] channel.c: Channel 0x7f0ca01129e0 'SIP/zvonobot-000000a1' allocated [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14433] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282856.394, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282856.394': 0x7f0c300fcf90 destroyed [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 From: ;tag=as22c76af6 To: ;tag=as509aa30f Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 389657146 389657146 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10944 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f2cd09b;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22c76af6 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[14433] res_stasis.c: calls_0: Subscribing to 213128 [Aug 18 10:34:17] DEBUG[14433] stasis/app.c: Channel '213128' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as509aa30f [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] DEBUG[14532] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] VERBOSE[14534] chan_sip.c: Audio is at 14784 [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000011', '', 'AppDial2', '(Outgoing Line)', 30, 0, 'BUSY', 3, '', '212978', '')] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14433] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14433] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14533] channel.c: Channel Announcer/ARI-00000032;1 setting write format path: gsm -> slin [Aug 18 10:34:17] DEBUG[14529] bridge_native_rtp.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14532] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] VERBOSE[14534] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Outgoing Call for 79821116912 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] VERBOSE[14535] chan_sip.c: Audio is at 10372 [Aug 18 10:34:17] VERBOSE[14535] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] VERBOSE[14535] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] VERBOSE[14535] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Initializing initreq for method INVITE - callid 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116912@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 3 [ 52]: From: ;tag=as20932b4d [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 6 [ 60]: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14535] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1997886737 1997886737 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Aug 18 10:34:17] DEBUG[14535] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[14534] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 389657146 389657146 IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[14529] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10944 RTP/AVP 0 8 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:17] DEBUG[14536] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[14536] http.c: HTTP Request URI is /ari/channels/212969/snoop?app=calls_0&spy=in [Aug 18 10:34:17] DEBUG[14529] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:17] DEBUG[14533] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:17] VERBOSE[14533] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 (Checking To) --From tag as22c76af6 --To-tag as509aa30f [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '248c25207a7726be43836ed9004043ec@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Strict routing enforced for session 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117004@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2877efca Max-Forwards: 70 From: ;tag=as22c76af6 To: ;tag=as509aa30f Contact: Call-ID: 248c25207a7726be43836ed9004043ec@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] VERBOSE[14534] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6282ms with no response [Aug 18 10:34:17] DEBUG[14536] http.c: match request [ari/channels/212969/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1c1f9bed1118a9cd31eace4006e214d8@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[14274] channel.c: Channel 0x7f0c80068db0 'SIP/zvonobot-00000083' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282857.395, detail: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14529] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 343394319 343394319 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18958 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.395': 0x7f0c300fcf90 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:454/channel:1629282857.395, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:454/channel:1629282857.395': 0x7f0c300fd040 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:454/channel:1629282857.395, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:454/channel:1629282857.395': 0x7f0c300fd040 destroyed [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282857.395, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.395': 0x7f0c300fcf90 destroyed [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000028', '', 'AppDial2', '(Outgoing Line)', 23, 0, 'BUSY', 3, '', '213004', '')] [Aug 18 10:34:17] DEBUG[14536] http.c: match request [ari/channels/212969/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14529] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] VERBOSE[14535] dial.c: Called zvonobot/79821116912 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282857.396, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.396': 0x7f0c300ab400 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:455/channel:1629282857.396, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:455/channel:1629282857.396': 0x7f0c30011650 created [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:455/channel:1629282857.396, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:455/channel:1629282857.396': 0x7f0c30011650 destroyed [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282857.396, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.396': 0x7f0c300ab400 destroyed [Aug 18 10:34:17] DEBUG[14536] http.c: match request [ari/channels/212969/snoop] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000077', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213080', '')] [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: Allocating new SIP dialog for 738fd11b612cb2a02e4676fb4d590539@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:17] DEBUG[14512] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c20064fb0' [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) RTP allocated port 11652 [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE creating session 0.0.0.0:11652 (11652) [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE create [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE add system candidates [Aug 18 10:34:17] DEBUG[14512] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:17] DEBUG[14512] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE add candidate: 159.65.48.104:11652, 2130706431 [Aug 18 10:34:17] DEBUG[14512] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:17] DEBUG[14512] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE add candidate: 10.131.0.10:11652, 2130706431 [Aug 18 10:34:17] DEBUG[14536] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[14529] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e is already using the new technology. [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Initializing initreq for method INVITE - callid 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116916@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 3 [ 52]: From: ;tag=as671c682b [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 6 [ 60]: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14534] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 171494189 171494189 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14784 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #77 [Aug 18 10:34:17] DEBUG[14534] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 From: ;tag=as2eb39fa6 To: ;tag=as7b46504f Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2eb39fa6 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7b46504f [Aug 18 10:34:17] VERBOSE[14534] dial.c: Called zvonobot/79821116916 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Finding handler for channels/212969/snoop [Aug 18 10:34:17] DEBUG[14512] rtp_engine.c: RTP instance '0x7f0c20064fb0' is setup and ready to go [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 (Checking To) --From tag as2eb39fa6 --To-tag as7b46504f [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (5) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1427440107 1427440107 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14706 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6216ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[14529] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: 0x7f0c1014bce0(SIP/zvonobot-00000026) is joining simple_bridge technology [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14285] channel.c: Channel 0x7f0c780382e0 'SIP/zvonobot-00000084' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 From: ;tag=as10d8c0eb To: ;tag=as70da9059 Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as10d8c0eb [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70da9059 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 (Checking To) --From tag as10d8c0eb --To-tag as70da9059 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '40cf8f1449337acf099d514511a8313d@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE stopped [Aug 18 10:34:17] DEBUG[14512] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:17] DEBUG[14512] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Finding handler for 212969 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channels create: Didn't match 212969 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14165] chan_sip.c: Hangup call SIP/zvonobot-0000007a, SIP callid 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14165] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:17] DEBUG[14165] res_rtp_asterisk.c: (0x7f0ca00fa8a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14165] res_rtp_asterisk.c: (0x7f0ca00fa8a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14165] channel.c: Channel 0x7f0ca0104fa0 'SIP/zvonobot-0000007a' destroying [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channels externalMedia: Didn't match 212969 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9 Max-Forwards: 70 From: ;tag=as64b58d1a To: ;tag=as0261f463 Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="259bd5c0", response="59f282fc5c7797e031e8ed9e3615fbc9" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14512] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:17] DEBUG[14536] res_ari.c: No explicit handler found for 212969. Using wildcard channelId. [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Finding handler for snoop [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as64b58d1a [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213088': is 0 interested in calls_0 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213088' unsubscribed from calls_0 [Aug 18 10:34:17] DEBUG[14431] channel.c: Channel 0x7f0c9c0d6ff0 'SIP/zvonobot-000000a2' allocated [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14431] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0261f463 [Aug 18 10:34:17] DEBUG[14539] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14165] stasis.c: Destroying topic. name: cache:300/channel:213088, detail: [Aug 18 10:34:17] DEBUG[14165] stasis.c: Topic 'cache:300/channel:213088': 0x7f0ca00ea4e0 destroyed [Aug 18 10:34:17] DEBUG[14165] stasis.c: Destroying topic. name: channel:213088, detail: [Aug 18 10:34:17] DEBUG[14165] stasis.c: Topic 'channel:213088': 0x7f0ca00de400 destroyed [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:17] DEBUG[14539] http.c: HTTP Request URI is /ari/channels/213088 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="259bd5c0", response="59f282fc5c7797e031e8ed9e3615fbc9" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:17] DEBUG[14539] http.c: match request [ari/channels/213088] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14539] http.c: match request [ari/channels/213088] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14539] http.c: match request [ari/channels/213088] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14539] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking From) --From tag as64b58d1a --To-tag as0261f463 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Finding handler for channels/213088 [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[14512] res_rtp_asterisk.c: (0x7f0c20064fb0) RTCP setup on RTP instance [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14431] res_stasis.c: calls_0: Subscribing to 213127 [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Finding handler for 213088 [Aug 18 10:34:17] DEBUG[14431] stasis/app.c: Channel '213127' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:17] DEBUG[14431] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14431] http.c: HTTP closing session. Top level [Aug 18 10:34:17] VERBOSE[14512] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking channels create: Didn't match 213088 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Outgoing Call for 79821116913 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:17] DEBUG[14512] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:17] DEBUG[14512] chan_sip.c: SIP call-id changed from '738fd11b612cb2a02e4676fb4d590539@127.0.1.1:5060' to '7496c2db547867277094cc9977c08d0c@159.65.48.104:5060' [Aug 18 10:34:17] DEBUG[14512] stasis.c: Creating topic. name: channel:213139, detail: [Aug 18 10:34:17] DEBUG[14512] stasis.c: Topic 'channel:213139': 0x7f0c2007dc20 created [Aug 18 10:34:17] DEBUG[14512] stasis.c: Creating topic. name: cache:456/channel:213139, detail: [Aug 18 10:34:17] DEBUG[14512] stasis.c: Topic 'cache:456/channel:213139': 0x7f0c200865b0 created [Aug 18 10:34:17] DEBUG[14539] res_ari.c: Checking channels externalMedia: Didn't match 213088 [Aug 18 10:34:17] DEBUG[14434] channel.c: Channel 0x7f0ca4122130 'SIP/zvonobot-000000a3' allocated [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14434] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:17] DEBUG[14539] res_ari.c: No explicit handler found for 213088. Using wildcard channelId. [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[13138] stasis/app.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e' is 2 interested in calls_0 [Aug 18 10:34:17] DEBUG[14525] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:17] DEBUG[14525] http.c: HTTP closing session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9;received=178.62.121.41 From: ;tag=as64b58d1a To: ;tag=as0261f463 Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:17] DEBUG[14434] res_stasis.c: calls_0: Subscribing to 213129 [Aug 18 10:34:17] DEBUG[14434] stasis/app.c: Channel '213129' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (1) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1997886737 1997886737 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (1) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 171494189 171494189 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14784 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session timer stopped: 42 - 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] DEBUG[14541] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[13941] bridge_channel.c: Setting 0x7f0c9403d460(SIP/zvonobot-0000001c) state from:0 to:1 [Aug 18 10:34:17] DEBUG[14541] http.c: HTTP Request URI is /ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/record?name=213003_eusdgnrhynNdoWptPwesQBAJOQYsAQjr&format=wav [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Outgoing Call for 79821116911 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:17] DEBUG[14434] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:17] DEBUG[14434] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14529] res_rtp_asterisk.c: (0x2c14110) RTP changing ssrc from 2067685933 to 1471776488 due to a source change [Aug 18 10:34:17] DEBUG[14541] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/record] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[13941] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pulling 0x7f0c9403d460(SIP/zvonobot-0000001c) [Aug 18 10:34:17] VERBOSE[13941] bridge_channel.c: Channel SIP/zvonobot-0000001c left 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:17] DEBUG[13941] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c9403d460(SIP/zvonobot-0000001c) is leaving simple_bridge technology [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:17] VERBOSE[14540] chan_sip.c: Audio is at 11634 [Aug 18 10:34:17] DEBUG[14541] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/record] with handler [phoneprov] len 9 [Aug 18 10:34:17] VERBOSE[14540] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] VERBOSE[14540] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] VERBOSE[14540] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[13941] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[13941] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[14541] http.c: match request [ari/bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/record] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[13941] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13941] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13941] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 From: ;tag=as7eb98fd0 To: ;tag=as30588395 Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7eb98fd0 [Aug 18 10:34:17] DEBUG[13941] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as30588395 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" [Aug 18 10:34:17] DEBUG[14541] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Initializing initreq for method INVITE - callid 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116913@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 3 [ 52]: From: ;tag=as1cc5f222 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 (Checking To) --From tag as7eb98fd0 --To-tag as30588395 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 6 [ 60]: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Destroying SIP dialog 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[13941] bridge_channel.c: Bridge is returning 0x7f0c9403d460(SIP/zvonobot-0000001c) to read format alaw [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca00fa8a0) DTLS stop [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca00fa8a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca00fa8a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca00fa8a0) ICE RTP transport deallocating [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca00fa8a0' [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8;received=159.65.48.104 From: ;tag=as0a05f417 To: ;tag=as029966c2 Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b16ead8" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Finding handler for bridges/cd02ec13-331f-440c-a360-837dbdfdba5e/record [Aug 18 10:34:17] DEBUG[13941] channel.c: Channel SIP/zvonobot-0000001c setting read format path: ulaw -> alaw [Aug 18 10:34:17] DEBUG[13941] bridge_channel.c: Bridge is returning 0x7f0c9403d460(SIP/zvonobot-0000001c) to write format alaw [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0a05f417 [Aug 18 10:34:17] DEBUG[14536] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14540] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #49 [Aug 18 10:34:17] DEBUG[14540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as029966c2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[13941] channel.c: Channel SIP/zvonobot-0000001c setting write format path: alaw -> ulaw [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[13941] stasis/control.c: 212991, 5fd3583d-12a2-4028-9389-fce6801ffb6b: Channel was departed from bridge [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[13941] stasis/app.c: bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b': is 1 interested in calls_0 [Aug 18 10:34:17] VERBOSE[14540] dial.c: Called zvonobot/79821116913 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[13047] stasis/control.c: 212991: Channel departing bridge [Aug 18 10:34:17] DEBUG[13047] bridge.c: Waiting for 0x7f0c9403d460(SIP/zvonobot-0000001c) bridge thread to die. [Aug 18 10:34:17] DEBUG[13941] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b16ead8" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] DEBUG[13047] stasis/app.c: channel '212991': is 1 interested in calls_0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[13047] channel.c: Channel 0x7f0c880272f0 'SIP/zvonobot-0000001c' hanging up. Refs: 3 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 (Checking To) --From tag as0a05f417 --To-tag as029966c2 [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Finding handler for cd02ec13-331f-440c-a360-837dbdfdba5e [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14541] res_ari.c: No explicit handler found for cd02ec13-331f-440c-a360-837dbdfdba5e. Using wildcard bridgeId. [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Finding handler for record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:17] DEBUG[14541] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:17] DEBUG[14541] stasis.c: Creating topic. name: channel:1629282857.398, detail: [Aug 18 10:34:17] VERBOSE[14542] chan_sip.c: Audio is at 17238 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14541] stasis.c: Topic 'channel:1629282857.398': 0x7f0cb4063810 created [Aug 18 10:34:17] DEBUG[14541] stasis.c: Creating topic. name: cache:457/channel:1629282857.398, detail: [Aug 18 10:34:17] DEBUG[14541] stasis.c: Topic 'cache:457/channel:1629282857.398': 0x7f0cb408d1a0 created [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000083 - start 1629282850.626799 answer 0.000000 end 1629282857.072522 dur 6.445 bill 1629282857.072 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000084 - start 1629282850.862238 answer 0.000000 end 1629282857.128753 dur 6.266 bill 1629282857.128 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282857.399, detail: [Aug 18 10:34:17] DEBUG[14161] chan_sip.c: Hangup call SIP/zvonobot-00000079, SIP callid 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14161] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:17] DEBUG[14161] res_rtp_asterisk.c: (0x7f0c9c093a50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14161] res_rtp_asterisk.c: (0x7f0c9c093a50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14161] channel.c: Channel 0x7f0c9c09ffc0 'SIP/zvonobot-00000079' destroying [Aug 18 10:34:17] VERBOSE[14542] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (4) INVITE - 5 [Aug 18 10:34:17] VERBOSE[14542] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213085': is 0 interested in calls_0 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213085' unsubscribed from calls_0 [Aug 18 10:34:17] DEBUG[14435] channel.c: Channel 0x7f0cb011fc00 'SIP/zvonobot-000000a4' allocated [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14435] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.399': 0x7f0c30011650 created [Aug 18 10:34:17] DEBUG[14544] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:458/channel:1629282857.399, detail: [Aug 18 10:34:17] VERBOSE[14542] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[14544] http.c: HTTP Request URI is /ari/channels/213085 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:458/channel:1629282857.399': 0x7f0c3013eba0 created [Aug 18 10:34:17] DEBUG[14544] http.c: match request [ari/channels/213085] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14435] res_stasis.c: calls_0: Subscribing to 213130 [Aug 18 10:34:17] DEBUG[14435] stasis/app.c: Channel '213130' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14435] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[14435] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14161] stasis.c: Destroying topic. name: cache:296/channel:213085, detail: [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:458/channel:1629282857.399, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:458/channel:1629282857.399': 0x7f0c3013eba0 destroyed [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282857.399, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.399': 0x7f0c30011650 destroyed [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:08', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000007a', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213088', '')] [Aug 18 10:34:17] DEBUG[14161] stasis.c: Topic 'cache:296/channel:213085': 0x7f0c9c0a27c0 destroyed [Aug 18 10:34:17] DEBUG[14161] stasis.c: Destroying topic. name: channel:213085, detail: [Aug 18 10:34:17] DEBUG[14161] stasis.c: Topic 'channel:213085': 0x7f0c9c0a1d40 destroyed [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12;received=159.65.48.104 From: ;tag=as42dc40dd To: ;tag=as0754e832 Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42dc40dd [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0754e832 [Aug 18 10:34:17] DEBUG[14544] http.c: match request [ari/channels/213085] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Initializing initreq for method INVITE - callid 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116911@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 3 [ 52]: From: ;tag=as2ff9bb68 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 6 [ 60]: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14542] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 671899043 671899043 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #69 [Aug 18 10:34:17] DEBUG[14542] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Outgoing Call for 79821116910 [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001c - start 1629282826.215841 answer 1629282843.422498 end 1629282857.235139 dur 31.019 bill 13.812 dispo ANSWERED [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282857.400, detail: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.400': 0x7f0c30011650 created [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 (Checking To) --From tag as42dc40dd --To-tag as0754e832 [Aug 18 10:34:17] DEBUG[14544] http.c: match request [ari/channels/213085] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14544] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '39299444695a01491d13a6704919adf8@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:17] VERBOSE[14542] dial.c: Called zvonobot/79821116911 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:459/channel:1629282857.400, detail: [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8800fb50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] VERBOSE[14545] chan_sip.c: Audio is at 15334 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8800fb50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117064@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20f02c12 Max-Forwards: 70 From: ;tag=as42dc40dd To: ;tag=as0754e832 Contact: Call-ID: 39299444695a01491d13a6704919adf8@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[14545] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 39299444695a01491d13a6704919adf8@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Finding handler for channels/213085 [Aug 18 10:34:17] DEBUG[14423] channel.c: Channel 0x7f0ca8109f70 'SIP/zvonobot-000000a5' allocated [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Destroying SIP dialog 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:17] VERBOSE[12942] dial.c: SIP/zvonobot-0000000c is busy [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c093a50) DTLS stop [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c093a50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[12942] channel.c: Channel 0x7f0c88017980 'SIP/zvonobot-0000000c' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c093a50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c093a50) ICE RTP transport deallocating [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14423] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9c093a50' [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:459/channel:1629282857.400': 0x7f0c3001ce70 created [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1997886737 1997886737 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (1) INVITE - 5 [Aug 18 10:34:17] DEBUG[14091] iostream.c: TCP socket error reading data: Connection reset by peer [Aug 18 10:34:17] DEBUG[13318] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[14091] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:17] DEBUG[14548] http.c: HTTP opening session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14548] http.c: HTTP Request URI is /ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:17] VERBOSE[14545] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[14548] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14548] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14548] http.c: match request [ari/bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14548] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Finding handler for bridges/aba705f1-c39f-408a-8a02-8c7f66ee7c7d/play [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:459/channel:1629282857.400, detail: [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 From: ;tag=as63ca65c0 To: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6159ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 171494189 171494189 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14784 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Finding handler for aba705f1-c39f-408a-8a02-8c7f66ee7c7d [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14548] res_ari.c: No explicit handler found for aba705f1-c39f-408a-8a02-8c7f66ee7c7d. Using wildcard bridgeId. [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Finding handler for play [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:17] DEBUG[14296] channel.c: Channel 0x7f0c7404c830 'SIP/zvonobot-00000085' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (1) INVITE - 5 [Aug 18 10:34:17] DEBUG[14548] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:459/channel:1629282857.400': 0x7f0c3001ce70 destroyed [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282857.400, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.400': 0x7f0c30011650 destroyed [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000079', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213085', '')] [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Finding handler for 213085 [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking channels create: Didn't match 213085 [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14544] res_ari.c: Checking channels externalMedia: Didn't match 213085 [Aug 18 10:34:17] DEBUG[14544] res_ari.c: No explicit handler found for 213085. Using wildcard channelId. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 671899043 671899043 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14423] res_stasis.c: calls_0: Subscribing to 213126 [Aug 18 10:34:17] DEBUG[14423] stasis/app.c: Channel '213126' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Outgoing Call for 79821116914 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] VERBOSE[14549] chan_sip.c: Audio is at 12208 [Aug 18 10:34:17] VERBOSE[14549] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] VERBOSE[14549] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[14423] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14423] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] VERBOSE[14549] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 From: ;tag=as10d8c0eb To: ;tag=as70da9059 Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[14545] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Initializing initreq for method INVITE - callid 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116910@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Initializing initreq for method INVITE - callid 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116914@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as10d8c0eb [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70da9059 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 3 [ 52]: From: ;tag=as288a5fb9 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 (Checking To) --From tag as10d8c0eb --To-tag as70da9059 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '40cf8f1449337acf099d514511a8313d@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 6 [ 60]: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 3 [ 52]: From: ;tag=as57d38e8b [Aug 18 10:34:17] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116961@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7 Max-Forwards: 70 From: ;tag=as10d8c0eb To: Contact: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 6 [ 60]: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14545] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116910@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 Max-Forwards: 70 From: ;tag=as57d38e8b To: Contact: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1404479865 1404479865 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Aug 18 10:34:17] DEBUG[14545] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6201ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] VERBOSE[14549] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116914@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 Max-Forwards: 70 From: ;tag=as288a5fb9 To: Contact: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2135045114 2135045114 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Aug 18 10:34:17] DEBUG[14549] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14297] channel.c: Channel 0x7f0c8c10a180 'SIP/zvonobot-00000086' hanging up. Refs: 2 [Aug 18 10:34:17] VERBOSE[14549] dial.c: Called zvonobot/79821116914 [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4be660db;received=159.65.48.104 From: ;tag=as5d2e4a10 To: ;tag=as01310a33 Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1470077576 1470077576 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 19176 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4be660db;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5d2e4a10 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as01310a33 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1470077576 1470077576 IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 19176 RTP/AVP 0 8 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 (Checking To) --From tag as5d2e4a10 --To-tag as01310a33 [Aug 18 10:34:17] VERBOSE[14545] dial.c: Called zvonobot/79821116910 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Got SDP version 1470077576 and unique parts [root 1470077576 IN IP4 178.62.121.41] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1470077576 1470077576 IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000000c - start 1629282824.061375 answer 0.000000 end 1629282857.388756 dur 33.327 bill 1629282857.388 dispo BUSY [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:17] DEBUG[14317] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[14317] http.c: HTTP closing session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) ICE set role failed; no ice instance [Aug 18 10:34:17] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) RTCP setting address on RTP instance [Aug 18 10:34:17] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c0801f290 -- Strict RTP learning after remote address set to: 178.62.121.41:19176 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:19176 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00dad48) from 0x7f0c147e2330 to 0x7f0c0801b7e8 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0063828) from 0x7f0c147e2330 to 0x7f0c0801b7e8 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb006f8a8) from 0x7f0c147e2330 to 0x7f0c0801b7e8 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) RTCP ignoring duplicate property [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000027 setting read format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000027 setting write format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) DTLS - ast_rtp_activate rtp=0x7f0c0801f290 - setup and perform DTLS' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801f290) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000085 - start 1629282851.152784 answer 0.000000 end 1629282857.451893 dur 6.299 bill 1629282857.451 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801f290) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Strict routing enforced for session 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117038@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK775440e3 Max-Forwards: 70 From: ;tag=as5d2e4a10 To: ;tag=as01310a33 Contact: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (4) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:17] VERBOSE[13141] dial.c: SIP/zvonobot-00000027 answered [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000086 - start 1629282851.195519 answer 0.000000 end 1629282857.530288 dur 6.334 bill 1629282857.530 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session timer started: 14 - 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 1768000ms [Aug 18 10:34:17] VERBOSE[13141] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000027 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:17] DEBUG[13141] stasis/app.c: Channel '213002' is 2 interested in calls_0 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:17] VERBOSE[13141] res_rtp_asterisk.c: 0x7f0c0801f290 -- Strict RTP switching to RTP target address 178.62.121.41:19176 as source [Aug 18 10:34:17] DEBUG[13141] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:17] DEBUG[13141] channel.c: Channel SIP/zvonobot-00000027 setting read format path: ulaw -> alaw [Aug 18 10:34:17] DEBUG[13141] channel.c: Channel SIP/zvonobot-00000027 setting write format path: alaw -> ulaw [Aug 18 10:34:17] DEBUG[14316] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[14316] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:17] DEBUG[14315] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[14315] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14552] http.c: HTTP opening session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 Max-Forwards: 70 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:17] DEBUG[14552] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:17] DEBUG[14552] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14552] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 [Aug 18 10:34:17] DEBUG[14552] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as31a963bc [Aug 18 10:34:17] DEBUG[14552] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as67678dc7 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking From) --From tag as31a963bc --To-tag as67678dc7 [Aug 18 10:34:17] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14552] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[14552] stasis.c: Creating topic. name: bridge:ef748304-0183-43ff-af76-f67b2d982f06, detail: [Aug 18 10:34:17] DEBUG[14552] stasis.c: Topic 'bridge:ef748304-0183-43ff-af76-f67b2d982f06': 0x7f0c2410d220 created [Aug 18 10:34:17] DEBUG[14552] stasis.c: Creating topic. name: cache:460/bridge:ef748304-0183-43ff-af76-f67b2d982f06, detail: [Aug 18 10:34:17] DEBUG[14552] stasis.c: Topic 'cache:460/bridge:ef748304-0183-43ff-af76-f67b2d982f06': 0x7f0c24112b50 created [Aug 18 10:34:17] DEBUG[14552] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[14552] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[14552] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14552] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:17] DEBUG[14552] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14552] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: calling simple_bridge technology constructor [Aug 18 10:34:17] DEBUG[14552] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: calling simple_bridge technology start [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14320] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[14320] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Setting 0x7f0c080248a0(SIP/zvonobot-00000012) state from:0 to:1 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7;received=178.62.121.41 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:17] DEBUG[14552] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14552] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (1) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116910@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 Max-Forwards: 70 From: ;tag=as57d38e8b To: Contact: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1404479865 1404479865 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14446] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[14446] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pulling 0x7f0c080248a0(SIP/zvonobot-00000012) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (1) INVITE - 5 [Aug 18 10:34:17] VERBOSE[13208] bridge_channel.c: Channel SIP/zvonobot-00000012 left 'softmix' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c080248a0(SIP/zvonobot-00000012) is leaving softmix technology [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Setting 0x7f0c2800f490(Announcer/ARI-00000009;2) state from:0 to:2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:17] DEBUG[14553] http.c: HTTP opening session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116914@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 Max-Forwards: 70 From: ;tag=as288a5fb9 To: Contact: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2135045114 2135045114 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 671899043 671899043 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session timer stopped: 15 - 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14553] http.c: HTTP Request URI is /ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/addChannel?channel=213002 [Aug 18 10:34:17] DEBUG[13208] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6'. Checking compatability for channels 'Announcer/ARI-00000009;2' and 'Recorder/ARI-00000003;2' [Aug 18 10:34:17] DEBUG[14418] channel.c: Channel 0x7f0c88058300 'SIP/zvonobot-000000a7' allocated [Aug 18 10:34:17] DEBUG[13208] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as could not get details [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14418] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13208] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] VERBOSE[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: switching from softmix technology to simple_bridge [Aug 18 10:34:17] DEBUG[14553] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology constructor [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c2800f490(Announcer/ARI-00000009;2) to dummy bridge temporarily [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: moving 0x7f0c100369e0(Recorder/ARI-00000003;2) to dummy bridge temporarily [Aug 18 10:34:17] DEBUG[14553] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2800f490(Announcer/ARI-00000009;2) is leaving softmix technology (dummy) [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is leaving softmix technology (dummy) [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology stop [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2800f490(Announcer/ARI-00000009;2) is joining simple_bridge technology [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Recorder/ARI-00000003;2 setting read format path: slin -> slin [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Announcer/ARI-00000009;2 setting write format path: slin -> slin [Aug 18 10:34:17] DEBUG[14553] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/addChannel] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Announcer/ARI-00000009;2 setting read format path: slin -> slin [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is joining simple_bridge technology [Aug 18 10:34:17] DEBUG[14418] res_stasis.c: calls_0: Subscribing to 213125 [Aug 18 10:34:17] DEBUG[14418] stasis/app.c: Channel '213125' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Recorder/ARI-00000003;2 setting read format path: slin -> slin [Aug 18 10:34:17] DEBUG[14418] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Announcer/ARI-00000009;2 setting write format path: slin -> slin [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000012 - start 1629282824.221431 answer 1629282830.240830 end 1629282857.693341 dur 33.471 bill 27.452 dispo ANSWERED [Aug 18 10:34:17] DEBUG[14418] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Announcer/ARI-00000009;2 setting read format path: slin -> slin [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Outgoing Call for 79821116915 [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel Recorder/ARI-00000003;2 setting write format path: slin -> slin [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[14553] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology start [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: deferring softmix technology destructor [Aug 18 10:34:17] DEBUG[14450] channel.c: Channel 0x7f0c2008f410 'Recorder/ARI-00000039;1' allocated [Aug 18 10:34:17] DEBUG[14450] stasis.c: Creating topic. name: channel:1629282857.401, detail: [Aug 18 10:34:17] DEBUG[14450] stasis.c: Topic 'channel:1629282857.401': 0x7f0c200b10f0 created [Aug 18 10:34:17] DEBUG[14450] stasis.c: Creating topic. name: cache:461/channel:1629282857.401, detail: [Aug 18 10:34:17] DEBUG[14450] stasis.c: Topic 'cache:461/channel:1629282857.401': 0x7f0c20085f10 created [Aug 18 10:34:17] DEBUG[13208] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: queueing action type:13 sub:1000 [Aug 18 10:34:17] DEBUG[14443] channel.c: Channel 0x2c522e0 'SIP/zvonobot-000000a8' allocated [Aug 18 10:34:17] DEBUG[13058] channel.c: Channel 0x7f0ca400e230 'Recorder/ARI-00000000;2' destroying [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 From: ;tag=as08e169d8 To: ;tag=as58d484f0 Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1044364027 1044364027 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12716 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:17] DEBUG[14216] channel.c: Channel 0x7f0c2419f2d0 'Announcer/ARI-00000038;1' allocated [Aug 18 10:34:17] DEBUG[14216] stasis.c: Creating topic. name: channel:1629282857.402, detail: [Aug 18 10:34:17] DEBUG[14216] stasis.c: Topic 'channel:1629282857.402': 0x7f0c24130800 created [Aug 18 10:34:17] DEBUG[14216] stasis.c: Creating topic. name: cache:462/channel:1629282857.402, detail: [Aug 18 10:34:17] DEBUG[14216] stasis.c: Topic 'cache:462/channel:1629282857.402': 0x7f0c24131220 created [Aug 18 10:34:17] WARNING[14349] app.c: No audio available on Recorder/ARI-0000002e;1?? [Aug 18 10:34:17] VERBOSE[13059] app.c: User hung up [Aug 18 10:34:17] DEBUG[13059] res_stasis_recording.c: 1629282827.30: Recording complete [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[14443] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] VERBOSE[14349] app.c: User hung up [Aug 18 10:34:17] DEBUG[14349] res_stasis_recording.c: 1629282847.267: Recording complete [Aug 18 10:34:17] DEBUG[14349] channel.c: Channel 0x7f0c80025890 'Recorder/ARI-0000002e;1' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[14443] res_stasis.c: calls_0: Subscribing to 213132 [Aug 18 10:34:17] DEBUG[14443] stasis/app.c: Channel '213132' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Outgoing Call for 79821116908 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] DEBUG[13059] channel.c: Channel 0x7f0ca4006700 'Recorder/ARI-00000000;1' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[14443] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14443] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08e169d8 [Aug 18 10:34:17] DEBUG[14331] app.c: One waitfor failed, trying another [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as58d484f0 [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Finding handler for bridges/ef748304-0183-43ff-af76-f67b2d982f06/addChannel [Aug 18 10:34:17] VERBOSE[14554] chan_sip.c: Audio is at 17970 [Aug 18 10:34:17] VERBOSE[14554] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] VERBOSE[14554] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:17] DEBUG[13058] stasis.c: Destroying topic. name: cache:39/channel:1629282827.31, detail: [Aug 18 10:34:17] DEBUG[13058] stasis.c: Topic 'cache:39/channel:1629282827.31': 0x7f0ca400ffc0 destroyed [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:17] DEBUG[13317] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: stopping mixing thread [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] VERBOSE[14554] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1044364027 1044364027 IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[13058] stasis.c: Destroying topic. name: channel:1629282827.31, detail: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12716 RTP/AVP 0 8 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:17] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[13058] stasis.c: Topic 'channel:1629282827.31': 0x7f0ca400e050 destroyed [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Initializing initreq for method INVITE - callid 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116915@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0 [Aug 18 10:34:17] DEBUG[13316] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pulling 0x7f0c2800f490(Announcer/ARI-00000009;2) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] VERBOSE[13316] bridge_channel.c: Channel Announcer/ARI-00000009;2 left 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 (Checking To) --From tag as08e169d8 --To-tag as58d484f0 [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Bridge is returning 0x7f0c080248a0(SIP/zvonobot-00000012) to read format alaw [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 3 [ 52]: From: ;tag=as66ca64f7 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:17] DEBUG[13316] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c2800f490(Announcer/ARI-00000009;2) is leaving simple_bridge technology [Aug 18 10:34:17] DEBUG[13316] bridge_channel.c: Setting 0x7f0c100369e0(Recorder/ARI-00000003;2) state from:0 to:2 [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel SIP/zvonobot-00000012 setting read format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Finding handler for ef748304-0183-43ff-af76-f67b2d982f06 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:17] DEBUG[13208] bridge_channel.c: Bridge is returning 0x7f0c080248a0(SIP/zvonobot-00000012) to write format alaw [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Got SDP version 1044364027 and unique parts [root 1044364027 IN IP4 178.62.121.41] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1044364027 1044364027 IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[13208] channel.c: Channel SIP/zvonobot-00000012 setting write format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[13208] stasis/control.c: 212983, 378d72c1-dd9d-472b-9f36-5c575a6102e6: Channel was departed from bridge [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[13208] stasis/app.c: bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6': is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:17] DEBUG[14553] res_ari.c: No explicit handler found for ef748304-0183-43ff-af76-f67b2d982f06. Using wildcard bridgeId. [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Finding handler for addChannel [Aug 18 10:34:17] DEBUG[13208] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:17] DEBUG[12968] stasis/control.c: 212983: Channel departing bridge [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:17] DEBUG[12968] bridge.c: Waiting for 0x7f0c080248a0(SIP/zvonobot-00000012) bridge thread to die. [Aug 18 10:34:17] DEBUG[12968] stasis/app.c: channel '212983': is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14553] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 6 [ 60]: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14554] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116915@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0 Max-Forwards: 70 From: ;tag=as66ca64f7 To: Contact: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 496785596 496785596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17970 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:34:17] DEBUG[14554] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[14554] dial.c: Called zvonobot/79821116915 [Aug 18 10:34:17] DEBUG[12968] channel.c: Channel 0x7f0c10025b50 'SIP/zvonobot-00000012' hanging up. Refs: 3 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:17] VERBOSE[14555] chan_sip.c: Audio is at 18670 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] VERBOSE[14555] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] DEBUG[14553] stasis/control.c: 213002: Sending channel add_to_bridge command [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) ICE set role failed; no ice instance [Aug 18 10:34:17] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) RTCP setting address on RTP instance [Aug 18 10:34:17] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c10011c80 -- Strict RTP learning after remote address set to: 178.62.121.41:12716 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12716 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb002b0c8) from 0x7f0c147e2330 to 0x7f0c1000e1d8 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0052848) from 0x7f0c147e2330 to 0x7f0c1000e1d8 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00e0588) from 0x7f0c147e2330 to 0x7f0c1000e1d8 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) RTCP ignoring duplicate property [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000002 setting read format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000002 setting write format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) DTLS - ast_rtp_activate rtp=0x7f0c10011c80 - setup and perform DTLS' [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10011c80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10011c80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] VERBOSE[14392] res_rtp_asterisk.c: 0x7f0c7800dea0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:11140 [Aug 18 10:34:17] DEBUG[13316] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:17] VERBOSE[14555] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:17] DEBUG[13316] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Strict routing enforced for session 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[13141] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000027 [Aug 18 10:34:17] DEBUG[13141] stasis/control.c: 213002: Adding to bridge ef748304-0183-43ff-af76-f67b2d982f06 [Aug 18 10:34:17] DEBUG[13141] stasis/app.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' is 1 interested in calls_0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] VERBOSE[14555] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] DEBUG[13316] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14560] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c1c0b2960(SIP/zvonobot-00000027) is joining [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117074@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK57b63e1a Max-Forwards: 70 From: ;tag=as08e169d8 To: ;tag=as58d484f0 Contact: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6382ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[14557] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[13316] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:17] VERBOSE[12874] dial.c: SIP/zvonobot-00000002 answered [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6231ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14560] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: pushing 0x7f0c1c0b2960(SIP/zvonobot-00000027) [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6226ms with no response [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14303] channel.c: Channel 0x7f0c38082e90 'SIP/zvonobot-00000087' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (3) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1997886737 1997886737 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Initializing initreq for method INVITE - callid 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6002ms with no response [Aug 18 10:34:17] VERBOSE[12874] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000002 [Aug 18 10:34:17] WARNING[20585] chan_sip.c: Hanging up call 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:17] DEBUG[14307] channel.c: Channel 0x7f0c940bd420 'SIP/zvonobot-00000088' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[12874] stasis/app.c: Channel '212966' is 2 interested in calls_0 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116908@178.62.121.41 SIP/2.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:17] VERBOSE[14560] bridge_channel.c: Channel SIP/zvonobot-00000027 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:17] DEBUG[13316] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14308] channel.c: Channel 0x7f0c900b6f50 'SIP/zvonobot-00000089' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[13316] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 is already using the new technology. [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:17] DEBUG[13210] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: pulling 0x7f0c100369e0(Recorder/ARI-00000003;2) [Aug 18 10:34:17] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:17] VERBOSE[13210] bridge_channel.c: Channel Recorder/ARI-00000003;2 left 'simple_bridge' stasis-bridge <378d72c1-dd9d-472b-9f36-5c575a6102e6> [Aug 18 10:34:17] DEBUG[13210] bridge_channel.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: 0x7f0c100369e0(Recorder/ARI-00000003;2) is leaving simple_bridge technology [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af [Aug 18 10:34:17] DEBUG[13210] bridge_native_rtp.c: Bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[20534] bridge_softmix.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: Waiting for mixing thread to die. [Aug 18 10:34:17] DEBUG[13210] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[13210] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[13210] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13210] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[13210] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6 is already using the new technology. [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 3 [ 52]: From: ;tag=as7dd13c21 [Aug 18 10:34:17] DEBUG[14318] channel.c: Channel 0x7f0c8413e2c0 'SIP/zvonobot-0000008b' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session timer started: 9 - 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 1768000ms [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 6 [ 60]: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:17 GMT [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] VERBOSE[14555] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116908@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af Max-Forwards: 70 From: ;tag=as7dd13c21 To: Contact: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1757417985 1757417985 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18670 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #108 [Aug 18 10:34:17] DEBUG[14555] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14557] http.c: HTTP Request URI is /ari/playbacks/29b62a77-22f1-4580-b914-d6cf9c0c52a7 [Aug 18 10:34:17] DEBUG[14561] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[13210] channel.c: Channel 0x7f0c10035e20 'Recorder/ARI-00000003;2' hanging up. Refs: 2 [Aug 18 10:34:17] VERBOSE[14555] dial.c: Called zvonobot/79821116908 [Aug 18 10:34:17] DEBUG[14561] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:17] DEBUG[14561] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14561] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14561] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64;received=159.65.48.104 From: ;tag=as3f810040 To: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37dd0a64;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f810040 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 35]: To: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 (Checking To) --From tag as3f810040 --To-tag [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 50439b5e4cdd97ff360cf8046b2cd5f5@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 515 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (3) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 171494189 171494189 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14784 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116910@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 Max-Forwards: 70 From: ;tag=as57d38e8b To: Contact: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1404479865 1404479865 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (2) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116914@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 Max-Forwards: 70 From: ;tag=as288a5fb9 To: Contact: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2135045114 2135045114 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a;received=159.65.48.104 From: ;tag=as3ecc0b7c To: ;tag=as6b227143 Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66504757" Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3ecc0b7c [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6b227143 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66504757" [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 (Checking To) --From tag as3ecc0b7c --To-tag as6b227143 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 From: ;tag=as5e4952fc To: ;tag=as228e10c3 Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1109970970 1109970970 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5e4952fc [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as228e10c3 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:17] DEBUG[14561] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:17] DEBUG[14557] http.c: match request [ari/playbacks/29b62a77-22f1-4580-b914-d6cf9c0c52a7] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] DEBUG[14557] http.c: match request [ari/playbacks/29b62a77-22f1-4580-b914-d6cf9c0c52a7] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14557] http.c: match request [ari/playbacks/29b62a77-22f1-4580-b914-d6cf9c0c52a7] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[13316] channel.c: Channel 0x7f0c28098390 'Announcer/ARI-00000009;2' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1109970970 1109970970 IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14956 RTP/AVP 0 8 101 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14561] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[14560] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:17] DEBUG[14557] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:17] DEBUG[14561] stasis.c: Creating topic. name: bridge:79f92216-f8f4-49dd-85f1-f154853e1fd1, detail: [Aug 18 10:34:17] DEBUG[14561] stasis.c: Topic 'bridge:79f92216-f8f4-49dd-85f1-f154853e1fd1': 0x7f0c3c11d670 created [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000087 - start 1629282851.335161 answer 0.000000 end 1629282857.814583 dur 6.479 bill 1629282857.814 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000089 - start 1629282851.528647 answer 0.000000 end 1629282857.816425 dur 6.287 bill 1629282857.816 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000088 - start 1629282851.450402 answer 0.000000 end 1629282857.816461 dur 6.366 bill 1629282857.816 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008b - start 1629282851.629244 answer 0.000000 end 1629282857.829965 dur 6.200 bill 1629282857.829 dispo NO ANSWER [Aug 18 10:34:17] DEBUG[14561] stasis.c: Creating topic. name: cache:463/bridge:79f92216-f8f4-49dd-85f1-f154853e1fd1, detail: [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 (Checking To) --From tag as5e4952fc --To-tag as228e10c3 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:17] DEBUG[14560] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Stopping retransmission on '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:17] DEBUG[14160] chan_sip.c: Hangup call SIP/zvonobot-00000078, SIP callid 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14160] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:17] DEBUG[14160] res_rtp_asterisk.c: (0x7f0c9408bf50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:17] DEBUG[14160] res_rtp_asterisk.c: (0x7f0c9408bf50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14160] channel.c: Channel 0x7f0c940b1960 'SIP/zvonobot-00000078' destroying [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Got SDP version 1109970970 and unique parts [root 1109970970 IN IP4 178.62.121.41] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1109970970 1109970970 IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[14561] stasis.c: Topic 'cache:463/bridge:79f92216-f8f4-49dd-85f1-f154853e1fd1': 0x7f0c3c12c710 created [Aug 18 10:34:17] DEBUG[14324] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:17] DEBUG[14561] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' can not use native RTP bridge as two channels are required [Aug 18 10:34:17] DEBUG[14561] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[14561] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14561] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:17] DEBUG[14561] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14561] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: calling simple_bridge technology constructor [Aug 18 10:34:17] DEBUG[14561] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: calling simple_bridge technology start [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Finding handler for playbacks/29b62a77-22f1-4580-b914-d6cf9c0c52a7 [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Finding handler for playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Finding handler for 29b62a77-22f1-4580-b914-d6cf9c0c52a7 [Aug 18 10:34:17] DEBUG[14557] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14557] res_ari.c: No explicit handler found for 29b62a77-22f1-4580-b914-d6cf9c0c52a7. Using wildcard playbackId. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:17] DEBUG[13318] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:17] DEBUG[14324] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] DEBUG[14560] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14565] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[14563] http.c: HTTP opening session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:17] DEBUG[14560] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[13318] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:17] DEBUG[14561] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] DEBUG[14561] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14560] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:17] DEBUG[14560] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06 is already using the new technology. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:17] DEBUG[14557] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:17] DEBUG[13318] channel.c: Channel Announcer/ARI-00000009;1 setting write format path: slin -> slin [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:17] DEBUG[14560] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c1c0b2960(SIP/zvonobot-00000027) is joining simple_bridge technology [Aug 18 10:34:17] NOTICE[13318] res_stasis_playback.c: 1629282832.89: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:17] DEBUG[13318] channel.c: Channel 0x7f0c280925d0 'Announcer/ARI-00000009;1' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[14564] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213084': is 0 interested in calls_0 [Aug 18 10:34:17] DEBUG[20620] stasis/app.c: channel '213084' unsubscribed from calls_0 [Aug 18 10:34:17] DEBUG[20620] stasis.c: Destroying topic. name: cache:295/channel:213084, detail: [Aug 18 10:34:17] DEBUG[20620] stasis.c: Topic 'cache:295/channel:213084': 0x7f0c940b4160 destroyed [Aug 18 10:34:17] DEBUG[20620] stasis.c: Destroying topic. name: channel:213084, detail: [Aug 18 10:34:17] DEBUG[20620] stasis.c: Topic 'channel:213084': 0x7f0c940b36e0 destroyed [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:17] DEBUG[14557] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:17] DEBUG[14392] res_rtp_asterisk.c: (0x7f0c7800c760) RTCP got report of 76 bytes from 178.62.121.41:11141 [Aug 18 10:34:17] DEBUG[14564] http.c: HTTP Request URI is /ari/channels/robot_212983 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14563] http.c: HTTP Request URI is /ari/channels/1629282835.133 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) ICE set role failed; no ice instance [Aug 18 10:34:17] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:17] DEBUG[14560] res_rtp_asterisk.c: (0x7f0c0801b610) RTP changing ssrc from 1408919060 to 888782111 due to a source change [Aug 18 10:34:17] DEBUG[14564] http.c: match request [ari/channels/robot_212983] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14564] http.c: match request [ari/channels/robot_212983] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14564] http.c: match request [ari/channels/robot_212983] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14564] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Finding handler for channels/robot_212983 [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Finding handler for robot_212983 [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking channels create: Didn't match robot_212983 [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14564] res_ari.c: Checking channels externalMedia: Didn't match robot_212983 [Aug 18 10:34:17] DEBUG[14564] res_ari.c: No explicit handler found for robot_212983. Using wildcard channelId. [Aug 18 10:34:17] DEBUG[13141] stasis/app.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' is 2 interested in calls_0 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) RTCP setting address on RTP instance [Aug 18 10:34:17] DEBUG[14553] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:17] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c700503c0 -- Strict RTP learning after remote address set to: 178.62.121.41:14956 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14956 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb011db38) from 0x7f0c147e2330 to 0x7f0c7004a038 [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb005afd8) from 0x7f0c147e2330 to 0x7f0c7004a038 [Aug 18 10:34:17] DEBUG[14553] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0054fd8) from 0x7f0c147e2330 to 0x7f0c7004a038 [Aug 18 10:34:17] DEBUG[14563] http.c: match request [ari/channels/1629282835.133] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) RTCP ignoring duplicate property [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282857.403, detail: [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003a setting read format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[14565] http.c: HTTP Request URI is /ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/addChannel?channel=212966 [Aug 18 10:34:17] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003a setting write format path: alaw -> alaw [Aug 18 10:34:17] DEBUG[14566] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[14565] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS - ast_rtp_activate rtp=0x7f0c700503c0 - setup and perform DTLS' [Aug 18 10:34:17] DEBUG[14563] http.c: match request [ari/channels/1629282835.133] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14565] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c700503c0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c700503c0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:17] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:17] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:17] DEBUG[14565] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/addChannel] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.403': 0x7f0c3001ce60 created [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Strict routing enforced for session 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14566] http.c: HTTP Request URI is /ari/channels/213084 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:17] DEBUG[14563] http.c: match request [ari/channels/1629282835.133] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117019@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f52815c Max-Forwards: 70 From: ;tag=as5e4952fc To: ;tag=as228e10c3 Contact: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #83 (6) INVITE - 5 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #83)) [Aug 18 10:34:17] DEBUG[14567] http.c: HTTP opening session. Top level [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116940@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed Max-Forwards: 70 From: ;tag=as0f1a808c To: Contact: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 840916179 840916179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11552 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[14565] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20545] stasis.c: Creating topic. name: cache:464/channel:1629282857.403, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:464/channel:1629282857.403': 0x7f0c301019c0 created [Aug 18 10:34:17] DEBUG[14566] http.c: match request [ari/channels/213084] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14566] http.c: match request [ari/channels/213084] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14566] http.c: match request [ari/channels/213084] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14566] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Finding handler for channels/213084 [Aug 18 10:34:17] DEBUG[14563] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[14567] http.c: HTTP Request URI is /ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/record?name=213002_SToxhQcamuWvHrRajlYeasVIVVuLUkoN&format=wav [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (1) INVITE - 5 [Aug 18 10:34:17] VERBOSE[13289] dial.c: SIP/zvonobot-0000003a answered [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116915@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0 Max-Forwards: 70 From: ;tag=as66ca64f7 To: Contact: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 496785596 496785596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17970 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Finding handler for bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/addChannel [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Session timer started: 127 - 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 1768000ms [Aug 18 10:34:17] VERBOSE[13289] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003a [Aug 18 10:34:17] DEBUG[13289] stasis/app.c: Channel '213021' is 2 interested in calls_0 [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: cache:464/channel:1629282857.403, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'cache:464/channel:1629282857.403': 0x7f0c301019c0 destroyed [Aug 18 10:34:17] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282857.403, detail: [Aug 18 10:34:17] DEBUG[20545] stasis.c: Topic 'channel:1629282857.403': 0x7f0c3001ce60 destroyed [Aug 18 10:34:17] DEBUG[14567] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/record] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000078', '', 'AppDial2', '(Outgoing Line)', 7, 0, 'NO ANSWER', 3, '', '213084', '')] [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:17] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Finding handler for 79f92216-f8f4-49dd-85f1-f154853e1fd1 [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Finding handler for 213084 [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14565] res_ari.c: No explicit handler found for 79f92216-f8f4-49dd-85f1-f154853e1fd1. Using wildcard bridgeId. [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking channels create: Didn't match 213084 [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Finding handler for addChannel [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Finding handler for channels/1629282835.133 [Aug 18 10:34:17] DEBUG[14565] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:17] DEBUG[14567] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/record] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14565] stasis/control.c: 212966: Sending channel add_to_bridge command [Aug 18 10:34:17] DEBUG[14568] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[14566] res_ari.c: Checking channels externalMedia: Didn't match 213084 [Aug 18 10:34:17] DEBUG[14568] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:17] DEBUG[14566] res_ari.c: No explicit handler found for 213084. Using wildcard channelId. [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Finding handler for channels [Aug 18 10:34:17] DEBUG[14568] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:17] DEBUG[14041] chan_sip.c: Hangup call SIP/zvonobot-0000006f, SIP callid 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14330] channel.c: Channel 0x7f0c2413d0a0 'SIP/zvonobot-00000096' allocated [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:17] DEBUG[12865] chan_sip.c: Hangup call SIP/zvonobot-00000000, SIP callid 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Finding handler for 1629282835.133 [Aug 18 10:34:17] DEBUG[14330] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:17] DEBUG[12865] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14351] channel.c: Channel 0x7f0c70042f30 'Snoop/213022-00000013' allocated [Aug 18 10:34:17] DEBUG[14041] res_rtp_asterisk.c: (0x7f0ca006da80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[12865] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14041] res_rtp_asterisk.c: (0x7f0ca006da80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14189] chan_sip.c: Hangup call SIP/zvonobot-0000007b, SIP callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14189] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:17] DEBUG[14189] res_rtp_asterisk.c: (0x7f0ca8078fc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14189] res_rtp_asterisk.c: (0x7f0ca8078fc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14189] channel.c: Channel 0x7f0ca80ecaf0 'SIP/zvonobot-0000007b' destroying [Aug 18 10:34:17] DEBUG[14459] channel.c: Channel 0x7f0c3c1372e0 'Announcer/ARI-0000003a;1' allocated [Aug 18 10:34:17] DEBUG[14459] stasis.c: Creating topic. name: channel:1629282857.404, detail: [Aug 18 10:34:17] DEBUG[14459] stasis.c: Topic 'channel:1629282857.404': 0x7f0c3c03cbc0 created [Aug 18 10:34:17] DEBUG[14459] stasis.c: Creating topic. name: cache:465/channel:1629282857.404, detail: [Aug 18 10:34:17] DEBUG[14459] stasis.c: Topic 'cache:465/channel:1629282857.404': 0x7f0c3c03d5b0 created [Aug 18 10:34:17] DEBUG[14567] http.c: match request [ari/bridges/ef748304-0183-43ff-af76-f67b2d982f06/record] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14041] channel.c: Channel 0x7f0ca00ed5f0 'SIP/zvonobot-0000006f' destroying [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking channels create: Didn't match 1629282835.133 [Aug 18 10:34:17] DEBUG[14146] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846'. Checking compatability for channels 'SIP/zvonobot-00000038' and 'Recorder/ARI-00000030;2' [Aug 18 10:34:17] VERBOSE[12865] chan_sip.c: Scheduling destruction of SIP dialog '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' in 6400 ms (Method: INVITE) [Aug 18 10:34:17] DEBUG[14146] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as channel 'SIP/zvonobot-00000038' has features which prevent it [Aug 18 10:34:17] DEBUG[14146] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[14146] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14146] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14146] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14146] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:17] DEBUG[14567] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[14340] channel.c: Channel 0x7f0c300a2d90 'Snoop/212965-00000012' allocated [Aug 18 10:34:17] DEBUG[14351] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9 Max-Forwards: 70 From: ;tag=as64b58d1a To: ;tag=as0261f463 Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="259bd5c0", response="59f282fc5c7797e031e8ed9e3615fbc9" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as64b58d1a [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0261f463 [Aug 18 10:34:17] DEBUG[14568] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14573] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[14351] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:17] DEBUG[14095] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa'. Checking compatability for channels 'SIP/zvonobot-00000001' and 'Recorder/ARI-0000002c;2' [Aug 18 10:34:17] DEBUG[14095] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' can not use native RTP bridge as channel 'SIP/zvonobot-00000001' has features which prevent it [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="259bd5c0", response="59f282fc5c7797e031e8ed9e3615fbc9" [Aug 18 10:34:17] DEBUG[14568] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14576] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:17] DEBUG[14095] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:17] DEBUG[14568] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[13076] bridge_channel.c: Setting 0x7f0c1c00dbc0(Snoop/212964-00000000) state from:0 to:1 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:17] DEBUG[14576] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213022&app=calls_0&format=slin16&external_host=127.0.0.1%3A50199 [Aug 18 10:34:17] DEBUG[14563] res_ari.c: Checking channels externalMedia: Didn't match 1629282835.133 [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:17] DEBUG[14330] res_stasis.c: calls_0: Subscribing to 213114 [Aug 18 10:34:17] DEBUG[14095] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14095] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:17] DEBUG[14573] http.c: HTTP Request URI is /ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play?media=sound%3Asilence%2F2 [Aug 18 10:34:17] DEBUG[14095] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:17] DEBUG[14095] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa is already using the new technology. [Aug 18 10:34:17] DEBUG[12865] chan_sip.c: Strict routing enforced for session 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[14567] res_ari.c: Finding handler for bridges/ef748304-0183-43ff-af76-f67b2d982f06/record [Aug 18 10:34:17] DEBUG[14573] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [httpstatus] len 10 [Aug 18 10:34:17] DEBUG[14340] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:17] VERBOSE[12865] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:17] DEBUG[12865] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[14573] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [phoneprov] len 9 [Aug 18 10:34:17] DEBUG[14573] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [ari] len 3 [Aug 18 10:34:17] DEBUG[14573] http.c: Match made with [ari] [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 (Checking From) --From tag as64b58d1a --To-tag as0261f463 [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:17] DEBUG[14340] http.c: HTTP closing session. Top level [Aug 18 10:34:17] DEBUG[14579] http.c: HTTP opening session. Top level [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:17] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:17] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:17] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:17] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1249a9e9;received=178.62.121.41 From: ;tag=as64b58d1a To: ;tag=as0261f463 Call-ID: 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 3782ef707142714164cf352b663534ff@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6442ms with no response [Aug 18 10:34:18] WARNING[20585] chan_sip.c: Hanging up call 3782ef707142714164cf352b663534ff@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3782ef707142714164cf352b663534ff@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14306] channel.c: Channel 0x7f0c8806dac0 'SIP/zvonobot-0000008a' hanging up. Refs: 2 [Aug 18 10:34:17] DEBUG[14568] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:17] DEBUG[12865] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] VERBOSE[12865] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:18] VERBOSE[12865] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: BYE sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41:5060", nonce="11c410aa", response="b5fe7439b49314fb4f0f18ebd5c2f549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[12865] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #130 [Aug 18 10:34:18] DEBUG[12865] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #108 (1) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #108)) [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Finding handler for bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116908@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af Max-Forwards: 70 From: ;tag=as7dd13c21 To: Contact: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1757417985 1757417985 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18670 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:17] DEBUG[14563] res_ari.c: No explicit handler found for 1629282835.133. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (3) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Session timer stopped: 16 - 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408bf50) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408bf50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408bf50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9408bf50) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9408bf50' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8078fc0) DTLS stop [Aug 18 10:34:18] DEBUG[13076] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: pulling 0x7f0c1c00dbc0(Snoop/212964-00000000) [Aug 18 10:34:18] VERBOSE[13076] bridge_channel.c: Channel Snoop/212964-00000000 left 'simple_bridge' stasis-bridge <87d87304-31e6-4326-b367-680423189269> [Aug 18 10:34:18] DEBUG[13076] bridge_channel.c: Bridge 87d87304-31e6-4326-b367-680423189269: 0x7f0c1c00dbc0(Snoop/212964-00000000) is leaving simple_bridge technology [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Finding handler for fbfe71c6-df7c-4b8c-8b66-37207aaf9846 [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14573] res_ari.c: No explicit handler found for fbfe71c6-df7c-4b8c-8b66-37207aaf9846. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Finding handler for play [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:18] DEBUG[14573] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8078fc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:17] DEBUG[14330] stasis/app.c: Channel '213114' is 1 interested in calls_0 [Aug 18 10:34:17] DEBUG[14567] res_ari.c: Finding handler for bridges [Aug 18 10:34:17] DEBUG[14576] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8078fc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8078fc0) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca8078fc0' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '40cf8f1449337acf099d514511a8313d@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca006da80) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca006da80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca006da80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca006da80) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca006da80' [Aug 18 10:34:18] DEBUG[14579] http.c: HTTP Request URI is /ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play?media=sound%3Asilence%2F2 [Aug 18 10:34:18] DEBUG[14568] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 Max-Forwards: 70 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[14584] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14584] http.c: HTTP Request URI is /ari/channels/213144?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116896&callerId=74950493843 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:18] DEBUG[14576] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14576] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14330] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:18] DEBUG[14576] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 [Aug 18 10:34:18] DEBUG[14573] stasis.c: Creating topic. name: channel:1629282858.405, detail: [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Finding handler for ef748304-0183-43ff-af76-f67b2d982f06 [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14567] res_ari.c: No explicit handler found for ef748304-0183-43ff-af76-f67b2d982f06. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Finding handler for record [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Outgoing Call for 79821116926 [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:18] VERBOSE[14585] chan_sip.c: Audio is at 15278 [Aug 18 10:34:18] DEBUG[14567] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:18] DEBUG[12874] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000002 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as31a963bc [Aug 18 10:34:17] DEBUG[14582] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14330] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14568] stasis.c: Creating topic. name: bridge:5c24e2ba-8671-4745-b349-4500db0d3cb5, detail: [Aug 18 10:34:18] DEBUG[12874] stasis/control.c: 212966: Adding to bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as67678dc7 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282858.406, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.406': 0x7f0c300ba000 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: cache:466/channel:1629282858.406, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:466/channel:1629282858.406': 0x7f0c30080030 created [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:18] DEBUG[14584] http.c: match request [ari/channels/213144] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking From) --From tag as31a963bc --To-tag as67678dc7 [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14573] stasis.c: Topic 'channel:1629282858.405': 0x7f0c8c06ade0 created [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:18] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14573] stasis.c: Creating topic. name: cache:467/channel:1629282858.405, detail: [Aug 18 10:34:18] DEBUG[14573] stasis.c: Topic 'cache:467/channel:1629282858.405': 0x7f0c8c118d00 created [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14579] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: cache:466/channel:1629282858.406, detail: [Aug 18 10:34:18] DEBUG[14568] stasis.c: Topic 'bridge:5c24e2ba-8671-4745-b349-4500db0d3cb5': 0x7f0c780ac5e0 created [Aug 18 10:34:18] DEBUG[14357] stasis.c: Creating topic. name: channel:1629282858.408, detail: [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] VERBOSE[14585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:18] DEBUG[14584] http.c: match request [ari/channels/213144] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:466/channel:1629282858.406': 0x7f0c30080030 destroyed [Aug 18 10:34:18] DEBUG[13076] bridge_native_rtp.c: Bridge '87d87304-31e6-4326-b367-680423189269' can not use native RTP bridge as two channels are required [Aug 18 10:34:18] DEBUG[14567] stasis.c: Creating topic. name: channel:1629282858.407, detail: [Aug 18 10:34:18] DEBUG[14584] http.c: match request [ari/channels/213144] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14579] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [phoneprov] len 9 [Aug 18 10:34:18] WARNING[14444] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000031;1 [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:18] DEBUG[14568] stasis.c: Creating topic. name: cache:468/bridge:5c24e2ba-8671-4745-b349-4500db0d3cb5, detail: [Aug 18 10:34:18] DEBUG[14568] stasis.c: Topic 'cache:468/bridge:5c24e2ba-8671-4745-b349-4500db0d3cb5': 0x7f0c78072780 created [Aug 18 10:34:18] DEBUG[14587] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7;received=178.62.121.41 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:18] DEBUG[14584] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (3) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 671899043 671899043 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[14582] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212965&app=calls_0&format=slin16&external_host=127.0.0.1%3A50065 [Aug 18 10:34:18] DEBUG[14588] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14568] bridge_native_rtp.c: Bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5' can not use native RTP bridge as two channels are required [Aug 18 10:34:18] DEBUG[14357] stasis.c: Topic 'channel:1629282858.408': 0x7f0c8c142b70 created [Aug 18 10:34:18] DEBUG[14357] stasis.c: Creating topic. name: cache:469/channel:1629282858.408, detail: [Aug 18 10:34:18] DEBUG[14357] stasis.c: Topic 'cache:469/channel:1629282858.408': 0x7f0c8c143820 created [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:18] DEBUG[14567] stasis.c: Topic 'channel:1629282858.407': 0x7f0c7c0a5470 created [Aug 18 10:34:18] DEBUG[13076] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:18] DEBUG[13076] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[13076] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[13076] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:18] DEBUG[13076] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269 is already using the new technology. [Aug 18 10:34:18] DEBUG[13076] bridge_channel.c: Bridge is returning 0x7f0c1c00dbc0(Snoop/212964-00000000) to read format slin [Aug 18 10:34:18] DEBUG[13076] channel.c: Channel Snoop/212964-00000000 setting read format path: slin -> slin [Aug 18 10:34:18] DEBUG[13076] bridge_channel.c: Bridge is returning 0x7f0c1c00dbc0(Snoop/212964-00000000) to write format slin [Aug 18 10:34:18] DEBUG[13076] channel.c: Channel Snoop/212964-00000000 setting write format path: slin -> slin [Aug 18 10:34:18] DEBUG[13076] stasis/control.c: 1629282827.33, 87d87304-31e6-4326-b367-680423189269: Channel was departed from bridge [Aug 18 10:34:18] VERBOSE[14585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:18] DEBUG[14567] stasis.c: Creating topic. name: cache:470/channel:1629282858.407, detail: [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14568] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:18] DEBUG[14576] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:18] DEBUG[14568] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[14568] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:18] DEBUG[14568] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:18] DEBUG[14568] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: calling simple_bridge technology constructor [Aug 18 10:34:18] DEBUG[14568] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: calling simple_bridge technology start [Aug 18 10:34:18] DEBUG[14568] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:18] DEBUG[14576] netsock2.c: Splitting '127.0.0.1:50199' into... [Aug 18 10:34:18] DEBUG[14576] netsock2.c: ...host '127.0.0.1' and port '50199'. [Aug 18 10:34:18] DEBUG[14568] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14582] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14584] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14576] netsock2.c: Splitting '127.0.0.1:50199' into... [Aug 18 10:34:18] DEBUG[14587] http.c: HTTP Request URI is /ari/channels/213145?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116895&callerId=74950493843 [Aug 18 10:34:18] DEBUG[14576] netsock2.c: ...host '127.0.0.1' and port '50199'. [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Finding handler for channels/213144 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282858.406, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.406': 0x7f0c300ba000 destroyed [Aug 18 10:34:18] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:08', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000007b', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213086', '')] [Aug 18 10:34:18] DEBUG[14579] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14582] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14569] stasis/app.c: Channel '1629282854.369' is 1 interested in calls_0 [Aug 18 10:34:18] DEBUG[14588] http.c: HTTP Request URI is /ari/channels/213086 [Aug 18 10:34:18] DEBUG[14589] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14576] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:18] DEBUG[14582] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14582] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14567] stasis.c: Topic 'cache:470/channel:1629282858.407': 0x7f0c7c087cf0 created [Aug 18 10:34:18] DEBUG[14570] stasis/app.c: Channel '1629282854.367' is 1 interested in calls_0 [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14576] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c880b67b0' [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) RTP allocated port 10534 [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) ICE creating session 127.0.0.1:10534 (10534) [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) ICE create [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) ICE add system candidates [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 From: ;tag=as64e6e544 To: ;tag=as189a4383 Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as64e6e544 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as189a4383 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 (Checking To) --From tag as64e6e544 --To-tag as189a4383 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Finding handler for 213144 [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking channels create: Didn't match 213144 [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14584] res_ari.c: Checking channels externalMedia: Didn't match 213144 [Aug 18 10:34:18] DEBUG[14584] res_ari.c: No explicit handler found for 213144. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[14579] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14576] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:18] DEBUG[14576] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) ICE add candidate: 159.65.48.104:10534, 2130706431 [Aug 18 10:34:18] DEBUG[14576] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:18] DEBUG[14576] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:18] DEBUG[14576] res_rtp_asterisk.c: (0x7f0c880b67b0) ICE add candidate: 10.131.0.10:10534, 2130706431 [Aug 18 10:34:18] DEBUG[14576] rtp_engine.c: RTP instance '0x7f0c880b67b0' is setup and ready to go [Aug 18 10:34:18] DEBUG[14576] stasis.c: Creating topic. name: channel:robot_213022, detail: [Aug 18 10:34:18] DEBUG[14576] stasis.c: Topic 'channel:robot_213022': 0x7f0c880c0020 created [Aug 18 10:34:18] DEBUG[14576] stasis.c: Creating topic. name: cache:471/channel:robot_213022, detail: [Aug 18 10:34:18] DEBUG[14576] stasis.c: Topic 'cache:471/channel:robot_213022': 0x7f0c880c4ac0 created [Aug 18 10:34:18] DEBUG[12874] stasis/app.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' is 1 interested in calls_0 [Aug 18 10:34:18] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:18] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[13076] stasis/app.c: bridge '87d87304-31e6-4326-b367-680423189269': is 2 interested in calls_0 [Aug 18 10:34:18] DEBUG[14589] http.c: HTTP Request URI is /ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/addChannel?channel=213021 [Aug 18 10:34:18] DEBUG[14569] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 156 instead [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Finding handler for bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Finding handler for e8b160c4-f3ae-46ad-bda0-ffa245693ffa [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14579] res_ari.c: No explicit handler found for e8b160c4-f3ae-46ad-bda0-ffa245693ffa. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Finding handler for play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:18] DEBUG[14579] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:18] DEBUG[14579] stasis.c: Creating topic. name: channel:1629282858.410, detail: [Aug 18 10:34:18] VERBOSE[14585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213086': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213086' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[14570] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 160 instead [Aug 18 10:34:18] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:18] DEBUG[13064] stasis/control.c: 1629282827.33: Channel departing bridge [Aug 18 10:34:18] DEBUG[14589] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14587] http.c: match request [ari/channels/213145] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14587] http.c: match request [ari/channels/213145] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] http.c: match request [ari/channels/213086] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[13064] bridge.c: Waiting for 0x7f0c1c00dbc0(Snoop/212964-00000000) bridge thread to die. [Aug 18 10:34:18] DEBUG[13076] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:18] DEBUG[14591] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:18] DEBUG[14589] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14596] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14589] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/addChannel] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] http.c: match request [ari/channels/213145] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14588] http.c: match request [ari/channels/213086] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14588] http.c: match request [ari/channels/213086] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14589] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14587] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14593] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c08021e90(SIP/zvonobot-00000002) is joining [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:18] DEBUG[14591] http.c: HTTP Request URI is /ari/channels/213149?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116891&callerId=74950493843 [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Finding handler for bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/addChannel [Aug 18 10:34:18] DEBUG[14579] stasis.c: Topic 'channel:1629282858.410': 0x7f0c94089760 created [Aug 18 10:34:18] DEBUG[14579] stasis.c: Creating topic. name: cache:472/channel:1629282858.410, detail: [Aug 18 10:34:18] DEBUG[14579] stasis.c: Topic 'cache:472/channel:1629282858.410': 0x7f0c940b0310 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282858.411, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.411': 0x7f0c30080030 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: cache:473/channel:1629282858.411, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:473/channel:1629282858.411': 0x7f0c30023840 created [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14582] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:18] DEBUG[14582] netsock2.c: Splitting '127.0.0.1:50065' into... [Aug 18 10:34:18] DEBUG[14582] netsock2.c: ...host '127.0.0.1' and port '50065'. [Aug 18 10:34:18] DEBUG[14582] netsock2.c: Splitting '127.0.0.1:50065' into... [Aug 18 10:34:18] DEBUG[14582] netsock2.c: ...host '127.0.0.1' and port '50065'. [Aug 18 10:34:18] DEBUG[14582] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:18] DEBUG[14582] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c900475d0' [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) RTP allocated port 14856 [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) ICE creating session 127.0.0.1:14856 (14856) [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) ICE create [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) ICE add system candidates [Aug 18 10:34:18] DEBUG[14582] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:18] DEBUG[14582] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) ICE add candidate: 159.65.48.104:14856, 2130706431 [Aug 18 10:34:18] DEBUG[14582] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:18] DEBUG[14582] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:18] DEBUG[14582] res_rtp_asterisk.c: (0x7f0c900475d0) ICE add candidate: 10.131.0.10:14856, 2130706431 [Aug 18 10:34:18] DEBUG[14582] rtp_engine.c: RTP instance '0x7f0c900475d0' is setup and ready to go [Aug 18 10:34:18] DEBUG[14582] stasis.c: Creating topic. name: channel:robot_212965, detail: [Aug 18 10:34:18] DEBUG[14582] stasis.c: Topic 'channel:robot_212965': 0x7f0c90073960 created [Aug 18 10:34:18] DEBUG[14582] stasis.c: Creating topic. name: cache:474/channel:robot_212965, detail: [Aug 18 10:34:18] DEBUG[14582] stasis.c: Topic 'cache:474/channel:robot_212965': 0x7f0c90057400 created [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 From: ;tag=as4d13c830 To: ;tag=as7af717bc Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[14588] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: cache:301/channel:213086, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'cache:301/channel:213086': 0x7f0ca8005f90 destroyed [Aug 18 10:34:18] DEBUG[14596] http.c: HTTP Request URI is /ari/channels/213150?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116890&callerId=74950493843 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Finding handler for channels/213145 [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Finding handler for 213145 [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking channels create: Didn't match 213145 [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14587] res_ari.c: Checking channels externalMedia: Didn't match 213145 [Aug 18 10:34:18] DEBUG[14587] res_ari.c: No explicit handler found for 213145. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d13c830 [Aug 18 10:34:18] DEBUG[14599] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Initializing initreq for method INVITE - callid 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14593] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: pushing 0x7f0c08021e90(SIP/zvonobot-00000002) [Aug 18 10:34:18] VERBOSE[14593] bridge_channel.c: Channel SIP/zvonobot-00000002 joined 'simple_bridge' stasis-bridge <79f92216-f8f4-49dd-85f1-f154853e1fd1> [Aug 18 10:34:18] DEBUG[14593] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' can not use native RTP bridge as two channels are required [Aug 18 10:34:18] DEBUG[14593] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:18] DEBUG[14593] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[14593] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[14593] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:18] DEBUG[14593] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1 is already using the new technology. [Aug 18 10:34:18] DEBUG[14593] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c08021e90(SIP/zvonobot-00000002) is joining simple_bridge technology [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: cache:473/channel:1629282858.411, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:473/channel:1629282858.411': 0x7f0c30023840 destroyed [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282858.411, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.411': 0x7f0c30080030 destroyed [Aug 18 10:34:18] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000006f', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213079', '')] [Aug 18 10:34:18] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[13064] stasis/app.c: channel '1629282827.33': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008a - start 1629282851.483885 answer 0.000000 end 1629282858.001503 dur 6.517 bill 1629282858.001 dispo NO ANSWER [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116926@178.62.121.41 SIP/2.0 [Aug 18 10:34:18] DEBUG[14593] res_rtp_asterisk.c: (0x7f0c1000e000) RTP changing ssrc from 2020082291 to 1406426276 due to a source change [Aug 18 10:34:18] DEBUG[14565] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:18] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7af717bc [Aug 18 10:34:18] DEBUG[14565] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 3 [ 52]: From: ;tag=as1885cc1f [Aug 18 10:34:18] DEBUG[14596] http.c: match request [ari/channels/213150] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14603] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:18] DEBUG[14591] http.c: match request [ari/channels/213149] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14591] http.c: match request [ari/channels/213149] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14591] http.c: match request [ari/channels/213149] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14591] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14591] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Finding handler for channels/213149 [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Finding handler for channels/213086 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14596] http.c: match request [ari/channels/213150] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14603] http.c: HTTP Request URI is /ari/channels/213152?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116888&callerId=74950493843 [Aug 18 10:34:18] DEBUG[14599] http.c: HTTP Request URI is /ari/channels/213151?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116889&callerId=74950493843 [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Finding handler for 5c24e2ba-8671-4745-b349-4500db0d3cb5 [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14589] res_ari.c: No explicit handler found for 5c24e2ba-8671-4745-b349-4500db0d3cb5. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Finding handler for addChannel [Aug 18 10:34:18] DEBUG[14589] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:18] DEBUG[14589] stasis/control.c: 213021: Sending channel add_to_bridge command [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[14603] http.c: match request [ari/channels/213152] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14603] http.c: match request [ari/channels/213152] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14603] http.c: match request [ari/channels/213152] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14603] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14596] http.c: match request [ari/channels/213150] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14596] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14596] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Finding handler for channels/213150 [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 6 [ 60]: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:18 GMT [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:18] VERBOSE[14585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1764277553 1764277553 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15278 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #135 [Aug 18 10:34:18] DEBUG[14585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] VERBOSE[14585] dial.c: Called zvonobot/79821116926 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 (Checking To) --From tag as4d13c830 --To-tag as7af717bc [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (2) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116915@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0 Max-Forwards: 70 From: ;tag=as66ca64f7 To: Contact: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 496785596 496785596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17970 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #130 (1) BYE - 8 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #130)) [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: BYE sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41:5060", nonce="11c410aa", response="b5fe7439b49314fb4f0f18ebd5c2f549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Finding handler for 213086 [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking channels create: Didn't match 213086 [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14588] res_ari.c: Checking channels externalMedia: Didn't match 213086 [Aug 18 10:34:18] DEBUG[14588] res_ari.c: No explicit handler found for 213086. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #108 (2) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #108)) [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[13064] stasis/app.c: channel '1629282827.33' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116908@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af Max-Forwards: 70 From: ;tag=as7dd13c21 To: Contact: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1757417985 1757417985 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18670 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[12874] stasis/app.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' is 2 interested in calls_0 [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14603] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14602] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[14569] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14602] http.c: HTTP Request URI is /ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/record?name=212966_moZBbkKqaNszlKGcZcxmnwJIRmMzjXTL&format=wav [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Finding handler for 213149 [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking channels create: Didn't match 213149 [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14591] res_ari.c: Checking channels externalMedia: Didn't match 213149 [Aug 18 10:34:18] DEBUG[14591] res_ari.c: No explicit handler found for 213149. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Finding handler for channels/213152 [Aug 18 10:34:18] DEBUG[14604] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Finding handler for 213150 [Aug 18 10:34:18] DEBUG[13064] channel.c: Channel 0x7f0c08011460 'Snoop/212964-00000000' hanging up. Refs: 3 [Aug 18 10:34:18] DEBUG[14604] http.c: HTTP Request URI is /ari/channels/213153?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116887&callerId=74950493843 [Aug 18 10:34:18] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14095] audiohook.c: Audiohook 0x7f0c3010f100 has stale audio in its factories. Flushing them both [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14604] http.c: match request [ari/channels/213153] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking channels create: Didn't match 213150 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] VERBOSE[14394] res_rtp_asterisk.c: 0x7f0c380281e0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:12658 [Aug 18 10:34:18] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 7 instead [Aug 18 10:34:18] DEBUG[14605] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14605] http.c: HTTP Request URI is /ari/channels/213148?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116892&callerId=74950493843 [Aug 18 10:34:18] DEBUG[14604] http.c: match request [ari/channels/213153] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14604] http.c: match request [ari/channels/213153] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14596] res_ari.c: Checking channels externalMedia: Didn't match 213150 [Aug 18 10:34:18] DEBUG[14596] res_ari.c: No explicit handler found for 213150. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: channel:213086, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'channel:213086': 0x7f0ca8077db0 destroyed [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213079': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213079' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: cache:281/channel:213079, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'cache:281/channel:213079': 0x7f0ca00f01b0 destroyed [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: channel:213079, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'channel:213079': 0x7f0ca00ef730 destroyed [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14605] http.c: match request [ari/channels/213148] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14604] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14604] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Finding handler for channels/213153 [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Finding handler for 213153 [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking channels create: Didn't match 213153 [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14604] res_ari.c: Checking channels externalMedia: Didn't match 213153 [Aug 18 10:34:18] DEBUG[14604] res_ari.c: No explicit handler found for 213153. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14599] http.c: match request [ari/channels/213151] with handler [httpstatus] len 10 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1070485752 1070485752 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15860 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[14606] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Finding handler for 213152 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking channels create: Didn't match 213152 [Aug 18 10:34:18] DEBUG[14596] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:18] DEBUG[14596] chan_sip.c: Allocating new SIP dialog for 5e16c53026cbb7381d1f9dd920f7e626@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:18] DEBUG[14596] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4085d30' [Aug 18 10:34:18] DEBUG[14599] http.c: match request [ari/channels/213151] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14599] http.c: match request [ari/channels/213151] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14602] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/record] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5f99fd9e [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[14605] http.c: match request [ari/channels/213148] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14606] http.c: HTTP Request URI is /ari/channels/213079 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14603] res_ari.c: Checking channels externalMedia: Didn't match 213152 [Aug 18 10:34:18] DEBUG[14603] res_ari.c: No explicit handler found for 213152. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14606] http.c: match request [ari/channels/213079] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14596] res_rtp_asterisk.c: (0x7f0cb4085d30) RTP allocated port 16040 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[14599] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14569] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1070485752 1070485752 IN IP4 178.62.121.41 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15860 RTP/AVP 0 8 101 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 (Checking To) --From tag as4ca0b4bd --To-tag as5f99fd9e [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Stopping retransmission on '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Strict routing enforced for session 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:18] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:18] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117037@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK304d74b2 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[14606] http.c: match request [ari/channels/213079] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 445 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 445 [Aug 18 10:34:18] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14606] http.c: match request [ari/channels/213079] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14605] http.c: match request [ari/channels/213148] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14596] res_rtp_asterisk.c: (0x7f0cb4085d30) ICE creating session 0.0.0.0:16040 (16040) [Aug 18 10:34:18] DEBUG[14602] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/record] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14602] http.c: match request [ari/bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/record] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14602] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Finding handler for bridges/79f92216-f8f4-49dd-85f1-f154853e1fd1/record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Finding handler for 79f92216-f8f4-49dd-85f1-f154853e1fd1 [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14602] res_ari.c: No explicit handler found for 79f92216-f8f4-49dd-85f1-f154853e1fd1. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Finding handler for record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:18] DEBUG[14602] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:18] DEBUG[14602] stasis.c: Creating topic. name: channel:1629282858.413, detail: [Aug 18 10:34:18] DEBUG[14602] stasis.c: Topic 'channel:1629282858.413': 0x7f0c1004f660 created [Aug 18 10:34:18] DEBUG[14602] stasis.c: Creating topic. name: cache:475/channel:1629282858.413, detail: [Aug 18 10:34:18] DEBUG[14602] stasis.c: Topic 'cache:475/channel:1629282858.413': 0x7f0c1004b000 created [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (3) INVITE - 5 [Aug 18 10:34:18] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:18] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14599] http.c: HTTP consuming request body [Aug 18 10:34:18] DEBUG[13935] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] WARNING[13682] logger: Log queue threshold (1000) exceeded. Discarding new messages. [Aug 18 10:34:18] DEBUG[14607] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14605] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:18] DEBUG[14596] res_rtp_asterisk.c: (0x7f0cb4085d30) ICE create [Aug 18 10:34:18] WARNING[20531] logger: Logging resumed. 944 messages discarded. [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Finding handler for channels/213093 [Aug 18 10:34:18] DEBUG[14623] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0075d58) from 0x7f0c147e2330 to 0x7f0c24007418 [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Finding handler for 9cefb3ad-33ea-4a52-96a1-42b677d6802c [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0116fb8) from 0x7f0c147e2330 to 0x7f0c24007418 [Aug 18 10:34:18] DEBUG[14623] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:18] DEBUG[14623] res_rtp_asterisk.c: (0x7f0c7005f720) ICE add candidate: 159.65.48.104:11544, 2130706431 [Aug 18 10:34:18] DEBUG[14623] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:18] DEBUG[14623] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:18] DEBUG[14623] res_rtp_asterisk.c: (0x7f0c7005f720) ICE add candidate: 10.131.0.10:11544, 2130706431 [Aug 18 10:34:18] DEBUG[14623] rtp_engine.c: RTP instance '0x7f0c7005f720' is setup and ready to go [Aug 18 10:34:18] DEBUG[14623] stasis.c: Creating topic. name: channel:robot_212967, detail: [Aug 18 10:34:18] DEBUG[14623] stasis.c: Topic 'channel:robot_212967': 0x7f0c70021be0 created [Aug 18 10:34:18] DEBUG[14623] stasis.c: Creating topic. name: cache:490/channel:robot_212967, detail: [Aug 18 10:34:18] DEBUG[14623] stasis.c: Topic 'cache:490/channel:robot_212967': 0x7f0c700131d0 created [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) RTCP ignoring duplicate property [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000014 setting read format path: alaw -> alaw [Aug 18 10:34:18] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000014 setting write format path: alaw -> alaw [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14620] res_ari.c: No explicit handler found for 9cefb3ad-33ea-4a52-96a1-42b677d6802c. Using wildcard bridgeId. [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Finding handler for 213093 [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking channels create: Didn't match 213093 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS - ast_rtp_activate rtp=0x7f0c2401dec0 - setup and perform DTLS' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2401dec0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2401dec0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:18] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:18] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:18] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Strict routing enforced for session 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Finding handler for play [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:18] DEBUG[14624] res_ari.c: Checking channels externalMedia: Didn't match 213093 [Aug 18 10:34:18] DEBUG[14624] res_ari.c: No explicit handler found for 213093. Using wildcard channelId. [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:18] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:18] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117056@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ee56735 Max-Forwards: 70 From: ;tag=as28b45d6b To: ;tag=as2a9101f2 Contact: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #130 (3) BYE - 8 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #130)) [Aug 18 10:34:18] DEBUG[14620] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: BYE sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41:5060", nonce="11c410aa", response="b5fe7439b49314fb4f0f18ebd5c2f549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] VERBOSE[12982] dial.c: SIP/zvonobot-00000014 answered [Aug 18 10:34:18] DEBUG[14381] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:18] DEBUG[13702] channel.c: Channel 0x7f0c2c096fd0 'Recorder/ARI-00000020;2' destroying [Aug 18 10:34:18] DEBUG[13177] chan_sip.c: Hangup call SIP/zvonobot-0000002a, SIP callid 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14381] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[13177] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (5) INVITE - 5 [Aug 18 10:34:18] DEBUG[13177] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] VERBOSE[12982] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000014 [Aug 18 10:34:18] DEBUG[13935] bridge_channel.c: Setting 0x7f0c20083ee0(Snoop/213007-0000000d) state from:0 to:1 [Aug 18 10:34:18] DEBUG[12982] stasis/app.c: Channel '212984' is 2 interested in calls_0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (4) INVITE - 5 [Aug 18 10:34:18] DEBUG[13702] stasis.c: Destroying topic. name: cache:220/channel:1629282839.185, detail: [Aug 18 10:34:18] DEBUG[13702] stasis.c: Topic 'cache:220/channel:1629282839.185': 0x7f0c2c0f3ac0 destroyed [Aug 18 10:34:18] DEBUG[13702] stasis.c: Destroying topic. name: channel:1629282839.185, detail: [Aug 18 10:34:18] DEBUG[13702] stasis.c: Topic 'channel:1629282839.185': 0x7f0c2c0f38f0 destroyed [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Session timer started: 83 - 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 1768000ms [Aug 18 10:34:18] DEBUG[14625] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14625] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:18] DEBUG[14625] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14625] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14625] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14625] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Finding handler for bridges [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:18] DEBUG[13935] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: pulling 0x7f0c20083ee0(Snoop/213007-0000000d) [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:18] DEBUG[14212] chan_sip.c: Hangup call SIP/zvonobot-00000081, SIP callid 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14212] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:18] VERBOSE[13935] bridge_channel.c: Channel Snoop/213007-0000000d left 'simple_bridge' stasis-bridge [Aug 18 10:34:18] DEBUG[14282] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:18] DEBUG[14143] channel.c: Channel 0x7f0c78095bd0 'Recorder/ARI-00000029;2' destroying [Aug 18 10:34:18] DEBUG[14282] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14205] chan_sip.c: Hangup call SIP/zvonobot-00000080, SIP callid 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14205] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:18] DEBUG[14212] res_rtp_asterisk.c: (0x7f0cb0134510) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14205] res_rtp_asterisk.c: (0x7f0c1012c700) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14212] res_rtp_asterisk.c: (0x7f0cb0134510) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14212] channel.c: Channel 0x7f0cb008f170 'SIP/zvonobot-00000081' destroying [Aug 18 10:34:18] DEBUG[14205] res_rtp_asterisk.c: (0x7f0c1012c700) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14204] chan_sip.c: Hangup call SIP/zvonobot-0000007e, SIP callid 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14204] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:18] DEBUG[14204] res_rtp_asterisk.c: (0x2c42620) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14204] res_rtp_asterisk.c: (0x2c42620) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14204] channel.c: Channel 0x2c47a50 'SIP/zvonobot-0000007e' destroying [Aug 18 10:34:18] DEBUG[14143] stasis.c: Destroying topic. name: cache:294/channel:1629282845.252, detail: [Aug 18 10:34:18] DEBUG[14143] stasis.c: Topic 'cache:294/channel:1629282845.252': 0x7f0c7803cb70 destroyed [Aug 18 10:34:18] DEBUG[14143] stasis.c: Destroying topic. name: channel:1629282845.252, detail: [Aug 18 10:34:18] DEBUG[14143] stasis.c: Topic 'channel:1629282845.252': 0x7f0c7803c690 destroyed [Aug 18 10:34:18] DEBUG[14205] channel.c: Channel 0x7f0c10131cc0 'SIP/zvonobot-00000080' destroying [Aug 18 10:34:18] DEBUG[13935] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c20083ee0(Snoop/213007-0000000d) is leaving simple_bridge technology [Aug 18 10:34:18] DEBUG[14625] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282858.430, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.430': 0x7f0c3009ffd0 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: cache:491/channel:1629282858.430, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:491/channel:1629282858.430': 0x7f0c300e62f0 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: cache:491/channel:1629282858.430, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:491/channel:1629282858.430': 0x7f0c300e62f0 destroyed [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282858.430, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.430': 0x7f0c3009ffd0 destroyed [Aug 18 10:34:18] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:09', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000081', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213091', '')] [Aug 18 10:34:18] DEBUG[14620] stasis.c: Creating topic. name: channel:1629282858.429, detail: [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213091': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213091' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213090': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213090' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: cache:307/channel:213090, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'cache:307/channel:213090': 0x2c23580 destroyed [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: channel:213090, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'channel:213090': 0x2c64460 destroyed [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282858.431, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.431': 0x7f0c300a0050 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: cache:492/channel:1629282858.431, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:492/channel:1629282858.431': 0x7f0c300fe540 created [Aug 18 10:34:18] DEBUG[14212] stasis.c: Destroying topic. name: cache:310/channel:213091, detail: [Aug 18 10:34:18] DEBUG[14212] stasis.c: Topic 'cache:310/channel:213091': 0x7f0cb011d9c0 destroyed [Aug 18 10:34:18] DEBUG[14212] stasis.c: Destroying topic. name: channel:213091, detail: [Aug 18 10:34:18] DEBUG[14212] stasis.c: Topic 'channel:213091': 0x7f0cb011d800 destroyed [Aug 18 10:34:18] DEBUG[14627] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14626] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: cache:492/channel:1629282858.431, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:492/channel:1629282858.431': 0x7f0c300fe540 destroyed [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282858.431, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.431': 0x7f0c300a0050 destroyed [Aug 18 10:34:18] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:09', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000007e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213090', '')] [Aug 18 10:34:18] DEBUG[14625] stasis.c: Creating topic. name: bridge:f495d952-07a0-4425-9378-2616afbaca10, detail: [Aug 18 10:34:18] DEBUG[14620] stasis.c: Topic 'channel:1629282858.429': 0x7f0c740acbb0 created [Aug 18 10:34:18] DEBUG[13935] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' can not use native RTP bridge as two channels are required [Aug 18 10:34:18] DEBUG[13935] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:18] DEBUG[13935] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[13935] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[13935] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:18] DEBUG[13935] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f is already using the new technology. [Aug 18 10:34:18] DEBUG[14626] http.c: HTTP Request URI is /ari/channels/213091 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213092': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis/app.c: channel '213092' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: cache:308/channel:213092, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'cache:308/channel:213092': 0x7f0c1004f3e0 destroyed [Aug 18 10:34:18] DEBUG[20620] stasis.c: Destroying topic. name: channel:213092, detail: [Aug 18 10:34:18] DEBUG[20620] stasis.c: Topic 'channel:213092': 0x7f0c101507e0 destroyed [Aug 18 10:34:18] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14627] http.c: HTTP Request URI is /ari/channels/213090 [Aug 18 10:34:18] DEBUG[14620] stasis.c: Creating topic. name: cache:493/channel:1629282858.429, detail: [Aug 18 10:34:18] DEBUG[14620] stasis.c: Topic 'cache:493/channel:1629282858.429': 0x7f0c74049f40 created [Aug 18 10:34:18] DEBUG[14627] http.c: match request [ari/channels/213090] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14628] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14626] http.c: match request [ari/channels/213091] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14486] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2;received=159.65.48.104 From: ;tag=as66bbd52b To: ;tag=as34030ed7 Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as66bbd52b [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34030ed7 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 (Checking To) --From tag as66bbd52b --To-tag as34030ed7 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Stopping retransmission on '7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[14486] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14628] http.c: HTTP Request URI is /ari/channels/213092 [Aug 18 10:34:18] DEBUG[14627] http.c: match request [ari/channels/213090] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14628] http.c: match request [ari/channels/213092] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117053@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bd0dbe2 Max-Forwards: 70 From: ;tag=as66bbd52b To: ;tag=as34030ed7 Contact: Call-ID: 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[14626] http.c: match request [ari/channels/213091] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14627] http.c: match request [ari/channels/213090] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14531] app.c: One waitfor failed, trying another [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14628] http.c: match request [ari/channels/213092] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[14625] stasis.c: Topic 'bridge:f495d952-07a0-4425-9378-2616afbaca10': 0x7f0c780701b0 created [Aug 18 10:34:18] VERBOSE[13011] dial.c: SIP/zvonobot-00000017 is busy [Aug 18 10:34:18] DEBUG[14627] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (4) INVITE - 5 [Aug 18 10:34:18] DEBUG[20545] cdr.c: Finalized CDR for Snoop/213007-0000000d - start 1629282842.192466 answer 1629282842.192466 end 1629282858.847310 dur 16.654 bill 16.654 dispo ANSWERED [Aug 18 10:34:18] DEBUG[13011] channel.c: Channel 0x7f0c3c020700 'SIP/zvonobot-00000017' hanging up. Refs: 2 [Aug 18 10:34:18] DEBUG[14628] http.c: match request [ari/channels/213092] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14625] stasis.c: Creating topic. name: cache:494/bridge:f495d952-07a0-4425-9378-2616afbaca10, detail: [Aug 18 10:34:18] DEBUG[14628] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 671899043 671899043 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17238 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (3) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1764277553 1764277553 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15278 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (5) INVITE - 5 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb404d3b0) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb404d3b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb404d3b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb404d3b0) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb404d3b0' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0134510) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0134510) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0134510) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0134510) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb0134510' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '41cffb51539db62640feb00322cb29ef@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x2c42620) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x2c42620) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x2c42620) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x2c42620) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x2c42620' [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1012c700) DTLS stop [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1012c700) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1012c700) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1012c700) ICE RTP transport deallocating [Aug 18 10:34:18] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c1012c700' [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 From: ;tag=as4d13c830 To: ;tag=as7af717bc Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d13c830 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7af717bc [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 (Checking To) --From tag as4d13c830 --To-tag as7af717bc [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 From: ;tag=as601f237f To: ;tag=as1ebdd48c Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as601f237f [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1ebdd48c [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:18] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:18] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 (Checking To) --From tag as601f237f --To-tag as1ebdd48c [Aug 18 10:34:18] DEBUG[13935] bridge_channel.c: Bridge is returning 0x7f0c20083ee0(Snoop/213007-0000000d) to read format slin [Aug 18 10:34:18] DEBUG[13935] channel.c: Channel Snoop/213007-0000000d setting read format path: slin -> slin [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Finding handler for channels/213092 [Aug 18 10:34:18] DEBUG[13935] bridge_channel.c: Bridge is returning 0x7f0c20083ee0(Snoop/213007-0000000d) to write format slin [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Finding handler for channels/213090 [Aug 18 10:34:18] DEBUG[14626] http.c: match request [ari/channels/213091] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14625] stasis.c: Topic 'cache:494/bridge:f495d952-07a0-4425-9378-2616afbaca10': 0x7f0c7804a7a0 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282858.432, detail: [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Stopping retransmission on '4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:18] DEBUG[13226] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[14625] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' can not use native RTP bridge as two channels are required [Aug 18 10:34:18] DEBUG[14625] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:18] DEBUG[14625] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:18] DEBUG[14625] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Finding handler for channels [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[14625] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[13935] channel.c: Channel Snoop/213007-0000000d setting write format path: slin -> slin [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:18] DEBUG[14625] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: calling simple_bridge technology constructor [Aug 18 10:34:18] DEBUG[14625] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: calling simple_bridge technology start [Aug 18 10:34:18] DEBUG[14625] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:18] DEBUG[13935] stasis/control.c: 1629282842.212, beb17a84-adfc-4fa3-b7a8-31977a540c1f: Channel was departed from bridge [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.432': 0x7f0c30067e40 created [Aug 18 10:34:18] DEBUG[14625] http.c: HTTP closing session. Top level [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4002ba40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Finding handler for 213090 [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Finding handler for 213092 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Creating topic. name: cache:495/channel:1629282858.432, detail: [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking channels create: Didn't match 213090 [Aug 18 10:34:18] DEBUG[14629] http.c: HTTP opening session. Top level [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[13935] stasis/app.c: bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f': is 3 interested in calls_0 [Aug 18 10:34:18] DEBUG[14627] res_ari.c: Checking channels externalMedia: Didn't match 213090 [Aug 18 10:34:18] DEBUG[14629] http.c: HTTP Request URI is /ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/addChannel?channel=212984 [Aug 18 10:34:18] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c4002ba40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117021@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709 Max-Forwards: 70 From: ;tag=as601f237f To: ;tag=as1ebdd48c Contact: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[14629] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking channels create: Didn't match 213092 [Aug 18 10:34:18] VERBOSE[13287] dial.c: SIP/zvonobot-00000039 is busy [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:18] DEBUG[13839] stasis/control.c: 1629282842.212: Channel departing bridge [Aug 18 10:34:18] DEBUG[14628] res_ari.c: Checking channels externalMedia: Didn't match 213092 [Aug 18 10:34:18] DEBUG[14628] res_ari.c: No explicit handler found for 213092. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[14629] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:18] DEBUG[13287] channel.c: Channel 0x7f0c40036ff0 'SIP/zvonobot-00000039' hanging up. Refs: 2 [Aug 18 10:34:18] DEBUG[14627] res_ari.c: No explicit handler found for 213090. Using wildcard channelId. [Aug 18 10:34:18] DEBUG[14629] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/addChannel] with handler [ari] len 3 [Aug 18 10:34:18] DEBUG[14629] http.c: Match made with [ari] [Aug 18 10:34:18] DEBUG[13935] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:18] DEBUG[13839] bridge.c: Waiting for 0x7f0c20083ee0(Snoop/213007-0000000d) bridge thread to die. [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 Max-Forwards: 70 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as31a963bc [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as67678dc7 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="08a4ff9b", response="7696d8cb6cd7eafac5b510cd34f1cd7c" [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 (Checking From) --From tag as31a963bc --To-tag as67678dc7 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:18] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:18] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:18] DEBUG[13839] stasis/app.c: channel '1629282842.212': is 0 interested in calls_0 [Aug 18 10:34:18] DEBUG[13839] stasis/app.c: channel '1629282842.212' unsubscribed from calls_0 [Aug 18 10:34:18] DEBUG[13839] channel.c: Channel 0x7f0c80065e60 'Snoop/213007-0000000d' hanging up. Refs: 3 [Aug 18 10:34:18] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:18] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:495/channel:1629282858.432': 0x7f0c300d5b80 created [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: cache:495/channel:1629282858.432, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'cache:495/channel:1629282858.432': 0x7f0c300d5b80 destroyed [Aug 18 10:34:18] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282858.432, detail: [Aug 18 10:34:18] DEBUG[20545] stasis.c: Topic 'channel:1629282858.432': 0x7f0c30067e40 destroyed [Aug 18 10:34:18] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:09', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000080', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213092', '')] [Aug 18 10:34:18] DEBUG[14626] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[12937] chan_sip.c: Hangup call SIP/zvonobot-0000000b, SIP callid 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[12937] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[12937] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[12937] channel.c: Channel 0x7f0c80011850 'SIP/zvonobot-0000000b' destroying [Aug 18 10:34:18] DEBUG[14629] res_ari.c: Finding handler for bridges/f495d952-07a0-4425-9378-2616afbaca10/addChannel [Aug 18 10:34:19] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel '212975': is 0 interested in calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel '212975' unsubscribed from calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: cache:18/channel:212975, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'cache:18/channel:212975': 0x7f0c80013510 destroyed [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: channel:212975, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'channel:212975': 0x7f0c80013310 destroyed [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000017 - start 1629282826.097593 answer 0.000000 end 1629282858.923487 dur 32.825 bill 1629282858.923 dispo BUSY [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000039 - start 1629282832.309357 answer 0.000000 end 1629282858.977158 dur 26.667 bill 1629282858.977 dispo BUSY [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282859.433, detail: [Aug 18 10:34:19] DEBUG[14493] channel.c: Channel 0x7f0cac090e50 'SIP/zvonobot-000000aa' allocated [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.433': 0x7f0c300d5b80 created [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: cache:496/channel:1629282859.433, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:496/channel:1629282859.433': 0x7f0c300d5aa0 created [Aug 18 10:34:19] DEBUG[14630] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14630] http.c: HTTP Request URI is /ari/channels/212975 [Aug 18 10:34:19] DEBUG[12959] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP 0x7f0ca000e480 -- Received packet from 178.62.121.41:15546, dropping due to strict RTP protection. [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Finding handler for channels/213091 [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14630] http.c: match request [ari/channels/212975] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: cache:496/channel:1629282859.433, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:496/channel:1629282859.433': 0x7f0c300d5aa0 destroyed [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282859.433, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.433': 0x7f0c300d5b80 destroyed [Aug 18 10:34:19] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000000b', '', 'AppDial2', '(Outgoing Line)', 32, 0, 'BUSY', 3, '', '212975', '')] [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Finding handler for channels [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14630] http.c: match request [ari/channels/212975] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Finding handler for f495d952-07a0-4425-9378-2616afbaca10 [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14629] res_ari.c: No explicit handler found for f495d952-07a0-4425-9378-2616afbaca10. Using wildcard bridgeId. [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Finding handler for addChannel [Aug 18 10:34:19] DEBUG[14629] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:19] DEBUG[14629] stasis/control.c: 212984: Sending channel add_to_bridge command [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14630] http.c: match request [ari/channels/212975] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14630] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14493] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Finding handler for channels/212975 [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Finding handler for channels [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Finding handler for 213091 [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking channels create: Didn't match 213091 [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:19] DEBUG[14626] res_ari.c: Checking channels externalMedia: Didn't match 213091 [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Finding handler for 212975 [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking channels create: Didn't match 212975 [Aug 18 10:34:19] DEBUG[14626] res_ari.c: No explicit handler found for 213091. Using wildcard channelId. [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14630] res_ari.c: Checking channels externalMedia: Didn't match 212975 [Aug 18 10:34:19] DEBUG[14497] channel.c: Channel 0x2c950d0 'SIP/zvonobot-000000ab' allocated [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14630] res_ari.c: No explicit handler found for 212975. Using wildcard channelId. [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14497] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK422b38f7;received=178.62.121.41 From: ;tag=as31a963bc To: ;tag=as67678dc7 Call-ID: 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14497] res_stasis.c: calls_0: Subscribing to 213137 [Aug 18 10:34:19] DEBUG[14497] stasis/app.c: Channel '213137' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14497] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (4) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116910@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 Max-Forwards: 70 From: ;tag=as57d38e8b To: Contact: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1404479865 1404479865 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14497] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (4) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116914@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 Max-Forwards: 70 From: ;tag=as288a5fb9 To: Contact: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2135045114 2135045114 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (5) INVITE - 5 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Outgoing Call for 79821116903 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #57)) [Aug 18 10:34:19] DEBUG[14493] res_stasis.c: calls_0: Subscribing to 213134 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116919@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK269403d2 Max-Forwards: 70 From: ;tag=as69045989 To: Contact: Call-ID: 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513281240 513281240 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15044 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (5) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116917@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788 Max-Forwards: 70 From: ;tag=as2489799b To: Contact: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1661788852 1661788852 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14493] stasis/app.c: Channel '213134' is 1 interested in calls_0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 From: ;tag=as15514e30 To: ;tag=as11c4e68e Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as15514e30 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as11c4e68e [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 (Checking To) --From tag as15514e30 --To-tag as11c4e68e [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[14493] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14493] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (3) INVITE - 5 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Outgoing Call for 79821116906 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116907@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772 Max-Forwards: 70 From: ;tag=as6c7cfd27 To: Contact: Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 680344710 680344710 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14511] channel.c: Channel 0x7f0c24139530 'SIP/zvonobot-000000ac' allocated [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Destroying SIP dialog 36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '36b08418271997272b9bd2346c9bd31a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS stop [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8000c760) ICE RTP transport deallocating [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8000c760' [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[14511] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] VERBOSE[14631] chan_sip.c: Audio is at 13978 [Aug 18 10:34:19] VERBOSE[14632] chan_sip.c: Audio is at 10106 [Aug 18 10:34:19] VERBOSE[14632] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[14631] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dff3038;received=159.65.48.104 From: ;tag=as1edcb3d8 To: ;tag=as0c855e05 Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 2083844308 2083844308 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17730 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dff3038;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1edcb3d8 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0c855e05 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 2083844308 2083844308 IN IP4 178.62.121.41 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17730 RTP/AVP 0 8 101 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:19] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 24 instead [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 (Checking To) --From tag as1edcb3d8 --To-tag as0c855e05 [Aug 18 10:34:19] VERBOSE[14632] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] VERBOSE[14631] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Stopping retransmission on '024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:19] VERBOSE[14632] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] VERBOSE[14631] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Got SDP version 2083844308 and unique parts [root 2083844308 IN IP4 178.62.121.41] [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 2083844308 2083844308 IN IP4 178.62.121.41... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14511] res_stasis.c: calls_0: Subscribing to 213141 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Initializing initreq for method INVITE - callid 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116906@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 3 [ 52]: From: ;tag=as5e0e197a [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 6 [ 60]: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14632] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116906@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74 Max-Forwards: 70 From: ;tag=as5e0e197a To: Contact: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 306689819 306689819 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #132 [Aug 18 10:34:19] DEBUG[14632] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14511] stasis/app.c: Channel '213141' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14511] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Initializing initreq for method INVITE - callid 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116903@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 3 [ 52]: From: ;tag=as41f91965 [Aug 18 10:34:19] DEBUG[14511] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 6 [ 60]: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Outgoing Call for 79821116899 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:19] VERBOSE[14632] dial.c: Called zvonobot/79821116906 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14631] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116903@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09 Max-Forwards: 70 From: ;tag=as41f91965 To: Contact: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1172096761 1172096761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #134 [Aug 18 10:34:19] DEBUG[14631] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:19] VERBOSE[14631] dial.c: Called zvonobot/79821116903 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18094150) ICE set role failed; no ice instance [Aug 18 10:34:19] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18094150) RTCP setting address on RTP instance [Aug 18 10:34:19] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c180a2570 -- Strict RTP learning after remote address set to: 178.62.121.41:17730 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:17730 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00473b8) from 0x7f0c147e2330 to 0x7f0c18094328 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00fb5f8) from 0x7f0c147e2330 to 0x7f0c18094328 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0139ad8) from 0x7f0c147e2330 to 0x7f0c18094328 [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18094150) RTCP ignoring duplicate property [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: Allocating new SIP dialog for 06cee45851a60dc84ac1014525477344@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:19] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003d setting read format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003d setting write format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18094150) DTLS - ast_rtp_activate rtp=0x7f0c180a2570 - setup and perform DTLS' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180a2570) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180a2570) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:19] DEBUG[14609] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c0c3c00' [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Strict routing enforced for session 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:19] VERBOSE[14633] chan_sip.c: Audio is at 17578 [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117016@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK05bb1b6e Max-Forwards: 70 From: ;tag=as1edcb3d8 To: ;tag=as0c855e05 Contact: Call-ID: 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #106 (5) INVITE - 5 [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) RTP allocated port 17200 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #106)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1300916893 1300916893 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14508 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE creating session 0.0.0.0:17200 (17200) [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE create [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE add system candidates [Aug 18 10:34:19] DEBUG[14609] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:19] DEBUG[14609] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE add candidate: 159.65.48.104:17200, 2130706431 [Aug 18 10:34:19] DEBUG[14609] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:19] DEBUG[14609] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE add candidate: 10.131.0.10:17200, 2130706431 [Aug 18 10:34:19] DEBUG[14609] rtp_engine.c: RTP instance '0x7f0c2c0c3c00' is setup and ready to go [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE stopped [Aug 18 10:34:19] DEBUG[14609] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:19] DEBUG[14609] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:19] VERBOSE[13411] dial.c: SIP/zvonobot-0000003d answered [Aug 18 10:34:19] VERBOSE[13411] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003d [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14609] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:19] DEBUG[13411] stasis/app.c: Channel '213024' is 2 interested in calls_0 [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 137837c51322c444587a45b5059337ee@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6401ms with no response [Aug 18 10:34:19] VERBOSE[14633] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Hanging up call 137837c51322c444587a45b5059337ee@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[13411] res_rtp_asterisk.c: 0x7f0c180a2570 -- Strict RTP switching to RTP target address 178.62.121.41:17730 as source [Aug 18 10:34:19] DEBUG[13411] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:19] DEBUG[13411] channel.c: Channel SIP/zvonobot-0000003d setting read format path: ulaw -> alaw [Aug 18 10:34:19] DEBUG[13411] channel.c: Channel SIP/zvonobot-0000003d setting write format path: alaw -> ulaw [Aug 18 10:34:19] DEBUG[14636] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14609] res_rtp_asterisk.c: (0x7f0c2c0c3c00) RTCP setup on RTP instance [Aug 18 10:34:19] DEBUG[14636] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:19] DEBUG[12982] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000014 [Aug 18 10:34:19] DEBUG[12982] stasis/control.c: 212984: Adding to bridge f495d952-07a0-4425-9378-2616afbaca10 [Aug 18 10:34:19] DEBUG[12982] stasis/app.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14372] channel.c: Channel 0x7f0c7c0a2360 'SIP/zvonobot-0000008c' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008c - start 1629282852.392901 answer 0.000000 end 1629282859.223828 dur 6.830 bill 1629282859.223 dispo NO ANSWER [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6424ms with no response [Aug 18 10:34:19] DEBUG[14636] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14636] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Hanging up call 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:19] VERBOSE[14633] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] DEBUG[14637] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0c20083ee0(SIP/zvonobot-00000014) is joining [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14636] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:19] DEBUG[14502] channel.c: Channel 0x7f0c080f4050 'SIP/zvonobot-000000af' allocated [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14502] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:19] DEBUG[14636] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008d - start 1629282852.481591 answer 0.000000 end 1629282859.241378 dur 6.759 bill 1629282859.241 dispo NO ANSWER [Aug 18 10:34:19] DEBUG[14377] channel.c: Channel 0x7f0c3c111a10 'SIP/zvonobot-0000008d' hanging up. Refs: 2 [Aug 18 10:34:19] VERBOSE[14609] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session timer started: 141 - 024bb77e2d9df3482194f20246a4de97@159.65.48.104:5060 1768000ms [Aug 18 10:34:19] DEBUG[14636] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:19] DEBUG[14636] stasis.c: Creating topic. name: bridge:724f7ab9-ed85-4748-9bb7-91218a7c6261, detail: [Aug 18 10:34:19] DEBUG[14636] stasis.c: Topic 'bridge:724f7ab9-ed85-4748-9bb7-91218a7c6261': 0x7f0c9802dde0 created [Aug 18 10:34:19] DEBUG[14636] stasis.c: Creating topic. name: cache:497/bridge:724f7ab9-ed85-4748-9bb7-91218a7c6261, detail: [Aug 18 10:34:19] DEBUG[14636] stasis.c: Topic 'cache:497/bridge:724f7ab9-ed85-4748-9bb7-91218a7c6261': 0x7f0c980128e0 created [Aug 18 10:34:19] DEBUG[14636] bridge_native_rtp.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[14636] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14636] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14636] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:19] DEBUG[14636] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14636] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling simple_bridge technology constructor [Aug 18 10:34:19] DEBUG[14636] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling simple_bridge technology start [Aug 18 10:34:19] DEBUG[14637] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: pushing 0x7f0c20083ee0(SIP/zvonobot-00000014) [Aug 18 10:34:19] VERBOSE[14637] bridge_channel.c: Channel SIP/zvonobot-00000014 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:19] DEBUG[14502] res_stasis.c: calls_0: Subscribing to 213140 [Aug 18 10:34:19] DEBUG[14502] stasis/app.c: Channel '213140' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14502] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14636] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14636] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Outgoing Call for 79821116900 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[14502] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14638] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14637] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[14637] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14637] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14637] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14638] http.c: HTTP Request URI is /ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/addChannel?channel=213024 [Aug 18 10:34:19] DEBUG[14637] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14637] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10 is already using the new technology. [Aug 18 10:34:19] VERBOSE[14633] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[14637] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0c20083ee0(SIP/zvonobot-00000014) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:19] DEBUG[14638] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14637] res_rtp_asterisk.c: (0x7f0c24007240) RTP changing ssrc from 929261269 to 336974372 due to a source change [Aug 18 10:34:19] DEBUG[12982] stasis/app.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' is 2 interested in calls_0 [Aug 18 10:34:19] DEBUG[14629] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:19] DEBUG[14638] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14638] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/addChannel] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14638] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Finding handler for bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/addChannel [Aug 18 10:34:19] DEBUG[14629] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Finding handler for 724f7ab9-ed85-4748-9bb7-91218a7c6261 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] DEBUG[14638] res_ari.c: No explicit handler found for 724f7ab9-ed85-4748-9bb7-91218a7c6261. Using wildcard bridgeId. [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Finding handler for addChannel [Aug 18 10:34:19] DEBUG[14638] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:19] DEBUG[14638] stasis/control.c: 213024: Sending channel add_to_bridge command [Aug 18 10:34:19] VERBOSE[14639] chan_sip.c: Audio is at 11616 [Aug 18 10:34:19] VERBOSE[14639] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6;received=159.65.48.104 From: ;tag=as0cd290ec To: ;tag=as01d9622d Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5ae8ae" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0cd290ec [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as01d9622d [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5ae8ae" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 (Checking To) --From tag as0cd290ec --To-tag as01d9622d [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #132 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #132)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116906@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74 Max-Forwards: 70 From: ;tag=as5e0e197a To: Contact: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 306689819 306689819 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116903@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09 Max-Forwards: 70 From: ;tag=as41f91965 To: Contact: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1172096761 1172096761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14640] http.c: HTTP opening session. Top level [Aug 18 10:34:19] VERBOSE[14639] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] DEBUG[14640] http.c: HTTP Request URI is /ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/record?name=212984_MMwrJXPbQmVTxOFiVLbLPNOJkQFoMEGA&format=wav [Aug 18 10:34:19] VERBOSE[14639] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14640] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/record] with handler [httpstatus] len 10 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 From: ;tag=as2d7c4d21 To: ;tag=as6fcc16b3 Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d7c4d21 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6fcc16b3 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 (Checking To) --From tag as2d7c4d21 --To-tag as6fcc16b3 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[14640] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/record] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14640] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/record] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (4) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116915@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0 Max-Forwards: 70 From: ;tag=as66ca64f7 To: Contact: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 496785596 496785596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17970 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14640] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Finding handler for bridges/f495d952-07a0-4425-9378-2616afbaca10/record [Aug 18 10:34:19] DEBUG[13411] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000003d [Aug 18 10:34:19] DEBUG[13411] stasis/control.c: 213024: Adding to bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261 [Aug 18 10:34:19] DEBUG[13411] stasis/app.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Finding handler for f495d952-07a0-4425-9378-2616afbaca10 [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14640] res_ari.c: No explicit handler found for f495d952-07a0-4425-9378-2616afbaca10. Using wildcard bridgeId. [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Finding handler for record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:19] DEBUG[14640] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:19] DEBUG[14640] stasis.c: Creating topic. name: channel:1629282859.434, detail: [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652 Max-Forwards: 70 From: ;tag=as4e94a883 To: ;tag=as181bb145 Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6ccf4cd6", response="3b85051dbb1a77b0db3c978d5568960d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as4e94a883 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as181bb145 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6ccf4cd6", response="3b85051dbb1a77b0db3c978d5568960d" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 (Checking From) --From tag as4e94a883 --To-tag as181bb145 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:19] DEBUG[14640] stasis.c: Topic 'channel:1629282859.434': 0x7f0cac00bd00 created [Aug 18 10:34:19] DEBUG[14640] stasis.c: Creating topic. name: cache:498/channel:1629282859.434, detail: [Aug 18 10:34:19] DEBUG[14640] stasis.c: Topic 'cache:498/channel:1629282859.434': 0x7f0cac06c570 created [Aug 18 10:34:19] DEBUG[14641] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) is joining [Aug 18 10:34:19] WARNING[14444] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000031;1 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Initializing initreq for method INVITE - callid 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116900@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 3 [ 52]: From: ;tag=as154d6f9d [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 6 [ 60]: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] DEBUG[14641] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: pushing 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) [Aug 18 10:34:19] VERBOSE[14639] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116900@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 Max-Forwards: 70 From: ;tag=as154d6f9d To: Contact: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2080919150 2080919150 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:19] VERBOSE[14641] bridge_channel.c: Channel SIP/zvonobot-0000003d joined 'simple_bridge' stasis-bridge <724f7ab9-ed85-4748-9bb7-91218a7c6261> [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #144 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652;received=178.62.121.41 From: ;tag=as4e94a883 To: ;tag=as181bb145 Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:19] DEBUG[14639] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[13347] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-00000008' [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] VERBOSE[14639] dial.c: Called zvonobot/79821116900 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session timer stopped: 7 - 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Setting 0x7f0c740224f0(SIP/zvonobot-00000008) state from:0 to:1 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 Max-Forwards: 70 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14609] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pulling 0x7f0c740224f0(SIP/zvonobot-00000008) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as20bcc5bd [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as31e40966 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[13347] bridge_channel.c: Channel SIP/zvonobot-00000008 left 'softmix' stasis-bridge [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c740224f0(SIP/zvonobot-00000008) is leaving softmix technology [Aug 18 10:34:19] DEBUG[14641] bridge_native_rtp.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Setting 0x7f0c2c0538b0(Announcer/ARI-00000018;2) state from:0 to:2 [Aug 18 10:34:19] DEBUG[14641] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Initializing initreq for method INVITE - callid 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116899@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 3 [ 52]: From: ;tag=as6a49d994 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 6 [ 60]: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14633] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116899@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 Max-Forwards: 70 From: ;tag=as6a49d994 To: Contact: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 370970940 370970940 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:34:19] DEBUG[14633] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking From) --From tag as20bcc5bd --To-tag as31e40966 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14641] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14641] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14641] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14641] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261 is already using the new technology. [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14641] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] DEBUG[14299] bridge_channel.c: Setting 0x7f0c80056c00(SIP/zvonobot-0000001f) state from:0 to:1 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09;received=178.62.121.41 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14299] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: pulling 0x7f0c80056c00(SIP/zvonobot-0000001f) [Aug 18 10:34:19] DEBUG[13347] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344'. Checking compatability for channels 'Announcer/ARI-00000018;2' and 'Recorder/ARI-0000000c;2' [Aug 18 10:34:19] DEBUG[13347] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as could not get details [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[13347] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] VERBOSE[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: switching from softmix technology to simple_bridge [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology constructor [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c2c0538b0(Announcer/ARI-00000018;2) to dummy bridge temporarily [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: moving 0x7f0c1c00f210(Recorder/ARI-0000000c;2) to dummy bridge temporarily [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c2c0538b0(Announcer/ARI-00000018;2) is leaving softmix technology (dummy) [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is leaving softmix technology (dummy) [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology stop [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c2c0538b0(Announcer/ARI-00000018;2) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Recorder/ARI-0000000c;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Announcer/ARI-00000018;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Announcer/ARI-00000018;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Recorder/ARI-0000000c;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Announcer/ARI-00000018;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Announcer/ARI-00000018;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel Recorder/ARI-0000000c;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology start [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: deferring softmix technology destructor [Aug 18 10:34:19] DEBUG[13347] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: queueing action type:13 sub:1000 [Aug 18 10:34:19] VERBOSE[14299] bridge_channel.c: Channel SIP/zvonobot-0000001f left 'simple_bridge' stasis-bridge <94ef4fc9-246b-4999-9567-b41f8ba44681> [Aug 18 10:34:19] DEBUG[14299] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c80056c00(SIP/zvonobot-0000001f) is leaving simple_bridge technology [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000008 - start 1629282822.246400 answer 1629282833.271079 end 1629282859.333871 dur 37.087 bill 26.062 dispo ANSWERED [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #108 (4) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #108)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116908@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af Max-Forwards: 70 From: ;tag=as7dd13c21 To: Contact: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1757417985 1757417985 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18670 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000001f - start 1629282828.030534 answer 1629282850.244476 end 1629282859.354117 dur 31.323 bill 9.109 dispo ANSWERED [Aug 18 10:34:19] VERBOSE[14633] dial.c: Called zvonobot/79821116899 [Aug 18 10:34:19] DEBUG[14299] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14299] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14299] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14299] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14299] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14510] channel.c: Channel 0x7f0c1807ed80 'SIP/zvonobot-000000ae' allocated [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14510] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[14299] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 is already using the new technology. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session timer stopped: 40 - 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14299] stasis/control.c: 212996, 94ef4fc9-246b-4999-9567-b41f8ba44681: Channel was departed from bridge [Aug 18 10:34:19] DEBUG[14641] res_rtp_asterisk.c: (0x7f0c18094150) RTP changing ssrc from 833708824 to 1377919390 due to a source change [Aug 18 10:34:19] DEBUG[14299] stasis/app.c: bridge '94ef4fc9-246b-4999-9567-b41f8ba44681': is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[13109] stasis/control.c: 212996: Channel departing bridge [Aug 18 10:34:19] DEBUG[13109] bridge.c: Waiting for 0x7f0c80056c00(SIP/zvonobot-0000001f) bridge thread to die. [Aug 18 10:34:19] DEBUG[14638] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:19] DEBUG[14638] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14299] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:19] DEBUG[13411] stasis/app.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261' is 2 interested in calls_0 [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:19] DEBUG[14643] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[13109] stasis/app.c: channel '212996': is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[13109] channel.c: Channel 0x7f0c84035ac0 'SIP/zvonobot-0000001f' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[14643] http.c: HTTP Request URI is /ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record?name=213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr&format=wav [Aug 18 10:34:19] DEBUG[14643] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14643] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14510] res_stasis.c: calls_0: Subscribing to 213135 [Aug 18 10:34:19] DEBUG[14643] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14510] stasis/app.c: Channel '213135' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14510] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14510] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14643] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Finding handler for bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Finding handler for 724f7ab9-ed85-4748-9bb7-91218a7c6261 [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14643] res_ari.c: No explicit handler found for 724f7ab9-ed85-4748-9bb7-91218a7c6261. Using wildcard bridgeId. [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Finding handler for record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:19] DEBUG[14643] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:19] DEBUG[14643] stasis.c: Creating topic. name: channel:1629282859.435, detail: [Aug 18 10:34:19] DEBUG[14643] stasis.c: Topic 'channel:1629282859.435': 0x2c26d40 created [Aug 18 10:34:19] DEBUG[14643] stasis.c: Creating topic. name: cache:499/channel:1629282859.435, detail: [Aug 18 10:34:19] DEBUG[14643] stasis.c: Topic 'cache:499/channel:1629282859.435': 0x2c26e70 created [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Outgoing Call for 79821116905 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] VERBOSE[14644] chan_sip.c: Audio is at 13604 [Aug 18 10:34:19] VERBOSE[14644] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[14644] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] VERBOSE[14644] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[14609] chan_sip.c: SIP call-id changed from '06cee45851a60dc84ac1014525477344@127.0.1.1:5060' to '1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060' [Aug 18 10:34:19] DEBUG[14609] stasis.c: Creating topic. name: channel:213147, detail: [Aug 18 10:34:19] DEBUG[14609] stasis.c: Topic 'channel:213147': 0x7f0c2c0ce6e0 created [Aug 18 10:34:19] DEBUG[14609] stasis.c: Creating topic. name: cache:500/channel:213147, detail: [Aug 18 10:34:19] DEBUG[14609] stasis.c: Topic 'cache:500/channel:213147': 0x7f0c2c0ad5e0 created [Aug 18 10:34:19] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Initializing initreq for method INVITE - callid 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116905@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 3 [ 52]: From: ;tag=as056b2e4e [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[13512] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: stopping mixing thread [Aug 18 10:34:19] DEBUG[13505] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pulling 0x7f0c2c0538b0(Announcer/ARI-00000018;2) [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] VERBOSE[13505] bridge_channel.c: Channel Announcer/ARI-00000018;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 6 [ 60]: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[13505] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c2c0538b0(Announcer/ARI-00000018;2) is leaving simple_bridge technology [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[13505] bridge_channel.c: Setting 0x7f0c1c00f210(Recorder/ARI-0000000c;2) state from:0 to:2 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 From: ;tag=as2ed109a6 To: ;tag=as7faa24c6 Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14644] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116905@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f Max-Forwards: 70 From: ;tag=as056b2e4e To: Contact: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1088879038 1088879038 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #96 [Aug 18 10:34:19] DEBUG[14644] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] VERBOSE[14644] dial.c: Called zvonobot/79821116905 [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Bridge is returning 0x7f0c740224f0(SIP/zvonobot-00000008) to read format alaw [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel SIP/zvonobot-00000008 setting read format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[13347] bridge_channel.c: Bridge is returning 0x7f0c740224f0(SIP/zvonobot-00000008) to write format alaw [Aug 18 10:34:19] DEBUG[13347] channel.c: Channel SIP/zvonobot-00000008 setting write format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[13347] stasis/control.c: 212972, d36cece3-ab54-488a-bcb0-0ed40691a344: Channel was departed from bridge [Aug 18 10:34:19] DEBUG[13347] stasis/app.c: bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344': is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[12900] stasis/control.c: 212972: Channel departing bridge [Aug 18 10:34:19] DEBUG[12900] bridge.c: Waiting for 0x7f0c740224f0(SIP/zvonobot-00000008) bridge thread to die. [Aug 18 10:34:19] DEBUG[13505] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[13505] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[13505] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14495] channel.c: Channel 0x7f0cb407fe90 'SIP/zvonobot-000000ad' allocated [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14495] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[13347] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:19] DEBUG[14262] chan_sip.c: Hangup call SIP/zvonobot-00000082, SIP callid 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14262] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:19] DEBUG[14262] res_rtp_asterisk.c: (0x7f0c7c018b60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14262] res_rtp_asterisk.c: (0x7f0c7c018b60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[14262] channel.c: Channel 0x7f0c7c013ef0 'SIP/zvonobot-00000082' destroying [Aug 18 10:34:19] DEBUG[13739] res_rtp_asterisk.c: (0x7f0ca804bf40) DTLS stop [Aug 18 10:34:19] DEBUG[13505] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[13739] res_rtp_asterisk.c: (0x7f0ca804bf40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[13739] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE RTP transport deallocating [Aug 18 10:34:19] DEBUG[13739] res_rtp_asterisk.c: (0x7f0ca804bf40) ICE stopped [Aug 18 10:34:19] DEBUG[13739] rtp_engine.c: Destroyed RTP instance '0x7f0ca804bf40' [Aug 18 10:34:19] DEBUG[13739] channel.c: Channel 0x7f0ca806b9f0 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40' destroying [Aug 18 10:34:19] DEBUG[13505] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[12900] stasis/app.c: channel '212972': is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[12900] channel.c: Channel 0x7f0c40010a50 'SIP/zvonobot-00000008' hanging up. Refs: 3 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ed109a6 [Aug 18 10:34:19] DEBUG[13505] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344 is already using the new technology. [Aug 18 10:34:19] DEBUG[14495] res_stasis.c: calls_0: Subscribing to 213136 [Aug 18 10:34:19] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:19] DEBUG[20534] bridge_softmix.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: Waiting for mixing thread to die. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7faa24c6 [Aug 18 10:34:19] DEBUG[13352] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: pulling 0x7f0c1c00f210(Recorder/ARI-0000000c;2) [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282859.437, detail: [Aug 18 10:34:19] VERBOSE[13352] bridge_channel.c: Channel Recorder/ARI-0000000c;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:19] DEBUG[13352] bridge_channel.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: 0x7f0c1c00f210(Recorder/ARI-0000000c;2) is leaving simple_bridge technology [Aug 18 10:34:19] DEBUG[14495] stasis/app.c: Channel '213136' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Outgoing Call for 79821116904 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[14495] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[13352] bridge_native_rtp.c: Bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[13352] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[13352] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[13352] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[13352] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[13352] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344 is already using the new technology. [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] VERBOSE[14646] chan_sip.c: Audio is at 13764 [Aug 18 10:34:19] VERBOSE[14646] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] DEBUG[14495] http.c: HTTP closing session. Top level [Aug 18 10:34:19] VERBOSE[14646] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[13505] channel.c: Channel 0x7f0c2c075a10 'Announcer/ARI-00000018;2' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.437': 0x7f0c30115050 created [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: cache:501/channel:1629282859.437, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:501/channel:1629282859.437': 0x7f0c301160b0 created [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: cache:501/channel:1629282859.437, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:501/channel:1629282859.437': 0x7f0c301160b0 destroyed [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282859.437, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.437': 0x7f0c30115050 destroyed [Aug 18 10:34:19] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:10', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000082', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213096', '')] [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 (Checking To) --From tag as2ed109a6 --To-tag as7faa24c6 [Aug 18 10:34:19] DEBUG[13352] channel.c: Channel 0x7f0c1c04bed0 'Recorder/ARI-0000000c;2' hanging up. Refs: 2 [Aug 18 10:34:19] VERBOSE[14646] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[14406] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:19] DEBUG[14406] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Initializing initreq for method INVITE - callid 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116904@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 3 [ 52]: From: ;tag=as1a5706e7 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 6 [ 60]: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14646] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116904@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face Max-Forwards: 70 From: ;tag=as1a5706e7 To: Contact: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 641212001 641212001 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13764 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #114 [Aug 18 10:34:19] DEBUG[14646] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[14646] dial.c: Called zvonobot/79821116904 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282859.438, detail: [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6451ms with no response [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.438': 0x7f0c300d5b80 created [Aug 18 10:34:19] DEBUG[20545] stasis.c: Creating topic. name: cache:502/channel:1629282859.438, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:502/channel:1629282859.438': 0x7f0c301160b0 created [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: cache:502/channel:1629282859.438, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'cache:502/channel:1629282859.438': 0x7f0c301160b0 destroyed [Aug 18 10:34:19] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282859.438, detail: [Aug 18 10:34:19] DEBUG[20545] stasis.c: Topic 'channel:1629282859.438': 0x7f0c300d5b80 destroyed [Aug 18 10:34:19] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:00', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213008', '')] [Aug 18 10:34:19] DEBUG[13537] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Hanging up call 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14379] channel.c: Channel 0x7f0c84147a20 'SIP/zvonobot-0000008e' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 465 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 465 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #144 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #144)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116900@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 Max-Forwards: 70 From: ;tag=as154d6f9d To: Contact: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2080919150 2080919150 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel '213096': is 0 interested in calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel '213096' unsubscribed from calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: cache:321/channel:213096, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'cache:321/channel:213096': 0x7f0c7c0c6230 destroyed [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: channel:213096, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'channel:213096': 0x7f0c7c015c40 destroyed [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel 'robot_213008': is 0 interested in calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis/app.c: channel 'robot_213008' unsubscribed from calls_0 [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: cache:222/channel:robot_213008, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'cache:222/channel:robot_213008': 0x7f0ca806e1b0 destroyed [Aug 18 10:34:19] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213008, detail: [Aug 18 10:34:19] DEBUG[20620] stasis.c: Topic 'channel:robot_213008': 0x7f0ca806d770 destroyed [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:19] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:19] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50291 - state 0 (Unknown) [Aug 18 10:34:19] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50291' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Aug 18 10:34:19] DEBUG[14649] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14649] http.c: HTTP Request URI is /ari/channels/213096 [Aug 18 10:34:19] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Recorder - ARI [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008e - start 1629282852.807782 answer 0.000000 end 1629282859.520502 dur 6.712 bill 1629282859.520 dispo NO ANSWER [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6437ms with no response [Aug 18 10:34:19] DEBUG[14649] http.c: match request [ari/channels/213096] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14650] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14650] http.c: HTTP Request URI is /ari/playbacks/2f05eea1-9012-4bac-a842-d4d3c70b926e [Aug 18 10:34:19] DEBUG[14650] http.c: match request [ari/playbacks/2f05eea1-9012-4bac-a842-d4d3c70b926e] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14650] http.c: match request [ari/playbacks/2f05eea1-9012-4bac-a842-d4d3c70b926e] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14650] http.c: match request [ari/playbacks/2f05eea1-9012-4bac-a842-d4d3c70b926e] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14650] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Finding handler for playbacks/2f05eea1-9012-4bac-a842-d4d3c70b926e [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Finding handler for playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14649] http.c: match request [ari/channels/213096] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14649] http.c: match request [ari/channels/213096] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14649] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Finding handler for channels/213096 [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Finding handler for channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Finding handler for 213096 [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking channels create: Didn't match 213096 [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14649] res_ari.c: Checking channels externalMedia: Didn't match 213096 [Aug 18 10:34:19] DEBUG[14649] res_ari.c: No explicit handler found for 213096. Using wildcard channelId. [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Finding handler for 2f05eea1-9012-4bac-a842-d4d3c70b926e [Aug 18 10:34:19] DEBUG[14650] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Hanging up call 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:19] DEBUG[14449] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14650] res_ari.c: No explicit handler found for 2f05eea1-9012-4bac-a842-d4d3c70b926e. Using wildcard playbackId. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:19] DEBUG[14652] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14652] http.c: HTTP Request URI is /ari/channels/robot_212972 [Aug 18 10:34:19] DEBUG[13537] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:19] DEBUG[13537] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:19] DEBUG[13537] channel.c: Channel Announcer/ARI-00000018;1 setting write format path: slin -> slin [Aug 18 10:34:19] NOTICE[13537] res_stasis_playback.c: 1629282835.138: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:19] DEBUG[13537] channel.c: Channel 0x7f0c2c070450 'Announcer/ARI-00000018;1' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[14650] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:19] DEBUG[14650] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14384] channel.c: Channel 0x7f0c3800da10 'SIP/zvonobot-0000008f' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[14652] http.c: match request [ari/channels/robot_212972] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14407] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:19] DEBUG[14407] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:19] DEBUG[14652] http.c: match request [ari/channels/robot_212972] with handler [phoneprov] len 9 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116899@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 Max-Forwards: 70 From: ;tag=as6a49d994 To: Contact: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 370970940 370970940 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000008f - start 1629282853.047477 answer 0.000000 end 1629282859.585994 dur 6.538 bill 1629282859.585 dispo NO ANSWER [Aug 18 10:34:19] DEBUG[14652] http.c: match request [ari/channels/robot_212972] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14652] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Finding handler for channels/robot_212972 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Finding handler for channels [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:19] DEBUG[14403] channel.c: Channel 0x7f0c4007cb10 'Recorder/ARI-00000036;2' allocated [Aug 18 10:34:19] DEBUG[14403] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] DEBUG[14200] channel.c: Channel 0x7f0ca010cb70 'Announcer/ARI-00000037;2' allocated [Aug 18 10:34:19] DEBUG[14200] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:19] DEBUG[14200] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000037;1' [Aug 18 10:34:19] DEBUG[14397] channel.c: Channel 0x7f0c3c149bd0 'Recorder/ARI-00000035;2' allocated [Aug 18 10:34:19] DEBUG[14524] channel.c: Channel 0x7f0c7c0a3d80 'Recorder/ARI-0000003b;1' allocated [Aug 18 10:34:19] DEBUG[14524] stasis.c: Creating topic. name: channel:1629282859.439, detail: [Aug 18 10:34:19] DEBUG[14524] stasis.c: Topic 'channel:1629282859.439': 0x7f0c7c00ec60 created [Aug 18 10:34:19] DEBUG[14524] stasis.c: Creating topic. name: cache:503/channel:1629282859.439, detail: [Aug 18 10:34:19] DEBUG[14524] stasis.c: Topic 'cache:503/channel:1629282859.439': 0x7f0c7c07d220 created [Aug 18 10:34:19] DEBUG[14432] channel.c: Channel 0x7f0c9c09b130 'Announcer/ARI-0000002f;1' destroying [Aug 18 10:34:19] DEBUG[14432] stasis.c: Destroying topic. name: cache:315/channel:1629282847.269, detail: [Aug 18 10:34:19] DEBUG[14432] stasis.c: Topic 'cache:315/channel:1629282847.269': 0x7f0c9c06dcb0 destroyed [Aug 18 10:34:19] DEBUG[14432] stasis.c: Destroying topic. name: channel:1629282847.269, detail: [Aug 18 10:34:19] DEBUG[14432] stasis.c: Topic 'channel:1629282847.269': 0x7f0c9c03a280 destroyed [Aug 18 10:34:19] DEBUG[14397] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] DEBUG[14304] channel.c: Channel 0x7f0c78069a80 'Recorder/ARI-00000034;2' allocated [Aug 18 10:34:19] DEBUG[14304] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #132 (2) INVITE - 5 [Aug 18 10:34:19] DEBUG[14405] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:19] DEBUG[14416] bridge_channel.c: Setting 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) state from:0 to:1 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:19] DEBUG[14654] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c780664b0(Recorder/ARI-00000034;2) is joining [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:19] DEBUG[14405] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #132)) [Aug 18 10:34:19] DEBUG[14655] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: 0x7f0ca00032d0(Announcer/ARI-00000037;2) is joining [Aug 18 10:34:19] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14653] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: 0x7f0c3c013500(Recorder/ARI-00000035;2) is joining [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116906@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74 Max-Forwards: 70 From: ;tag=as5e0e197a To: Contact: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 306689819 306689819 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:19] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Finding handler for robot_212972 [Aug 18 10:34:19] DEBUG[14656] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40070f90(Recorder/ARI-00000036;2) is joining [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (2) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking channels create: Didn't match robot_212972 [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:19] DEBUG[14416] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pulling 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116903@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09 Max-Forwards: 70 From: ;tag=as41f91965 To: Contact: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1172096761 1172096761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] VERBOSE[14416] bridge_channel.c: Channel Announcer/ARI-0000002f;2 left 'softmix' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:19] DEBUG[14652] res_ari.c: Checking channels externalMedia: Didn't match robot_212972 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14416] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c9c0f0540(Announcer/ARI-0000002f;2) is leaving softmix technology [Aug 18 10:34:19] DEBUG[14652] res_ari.c: No explicit handler found for robot_212972. Using wildcard channelId. [Aug 18 10:34:19] DEBUG[14654] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: pushing 0x7f0c780664b0(Recorder/ARI-00000034;2) [Aug 18 10:34:19] DEBUG[14654] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6458ms with no response [Aug 18 10:34:19] VERBOSE[14654] bridge_channel.c: Channel Recorder/ARI-00000034;2 joined 'simple_bridge' stasis-bridge <94ef4fc9-246b-4999-9567-b41f8ba44681> [Aug 18 10:34:19] DEBUG[14655] bridge_channel.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: pushing 0x7f0ca00032d0(Announcer/ARI-00000037;2) [Aug 18 10:34:19] WARNING[20585] chan_sip.c: Hanging up call 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14656] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: pushing 0x7f0c40070f90(Recorder/ARI-00000036;2) [Aug 18 10:34:19] DEBUG[14386] channel.c: Channel 0x7f0c80074680 'SIP/zvonobot-00000091' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000091 - start 1629282853.127676 answer 0.000000 end 1629282859.656303 dur 6.528 bill 1629282859.656 dispo NO ANSWER [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:19] DEBUG[14416] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677'. Checking compatability for channels 'SIP/zvonobot-00000013' and 'Recorder/ARI-00000024;2' [Aug 18 10:34:19] DEBUG[14654] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:19] DEBUG[14656] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] VERBOSE[14656] bridge_channel.c: Channel Recorder/ARI-00000036;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:19] DEBUG[14654] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14654] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14654] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14654] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14654] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 is already using the new technology. [Aug 18 10:34:19] DEBUG[14654] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c780664b0(Recorder/ARI-00000034;2) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:19] DEBUG[14416] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as channel 'SIP/zvonobot-00000013' has features which prevent it [Aug 18 10:34:19] DEBUG[14304] res_stasis_recording.c: 1629282851.317: Sending record(212996_tETNWgXOdyOJtmtuSqFlpvUqMEjxiyua.wav) command [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263;received=159.65.48.104 From: ;tag=as7d784780 To: ;tag=as76a7304d Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a209b2b" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d784780 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as76a7304d [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a209b2b" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 (Checking To) --From tag as7d784780 --To-tag as76a7304d [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #130 (4) BYE - 8 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #130)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: BYE sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41:5060", nonce="11c410aa", response="b5fe7439b49314fb4f0f18ebd5c2f549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116905@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f Max-Forwards: 70 From: ;tag=as056b2e4e To: Contact: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1088879038 1088879038 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (1) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116904@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face Max-Forwards: 70 From: ;tag=as1a5706e7 To: Contact: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 641212001 641212001 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13764 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #144 (2) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #144)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116900@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 Max-Forwards: 70 From: ;tag=as154d6f9d To: Contact: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2080919150 2080919150 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (2) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116899@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 Max-Forwards: 70 From: ;tag=as6a49d994 To: Contact: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 370970940 370970940 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (4) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1764277553 1764277553 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15278 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14653] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: pushing 0x7f0c3c013500(Recorder/ARI-00000035;2) [Aug 18 10:34:19] DEBUG[14304] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14416] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] VERBOSE[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: switching from softmix technology to simple_bridge [Aug 18 10:34:19] DEBUG[14304] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology constructor [Aug 18 10:34:19] DEBUG[14656] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759'. Checking compatability for channels 'SIP/zvonobot-00000029' and 'Recorder/ARI-00000036;2' [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c18091350(SIP/zvonobot-00000013) to dummy bridge temporarily [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c940389d0(Recorder/ARI-00000024;2) to dummy bridge temporarily [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is leaving softmix technology (dummy) [Aug 18 10:34:19] DEBUG[14656] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' can not use native RTP bridge as could not get details [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is leaving softmix technology (dummy) [Aug 18 10:34:19] DEBUG[14656] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology stop [Aug 18 10:34:19] DEBUG[14656] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14656] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14656] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14656] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759 is already using the new technology. [Aug 18 10:34:19] DEBUG[14658] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14514] channel.c: Channel 0x7f0c280eeac0 'SIP/zvonobot-000000b1' allocated [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14514] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[14656] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40070f90(Recorder/ARI-00000036;2) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Destroying SIP dialog 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14658] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:19] DEBUG[14416] channel.c: Channel Recorder/ARI-00000024;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c018b60) DTLS stop [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c018b60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c018b60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c018b60) ICE RTP transport deallocating [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7c018b60' [Aug 18 10:34:19] DEBUG[14658] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:19] DEBUG[14514] res_stasis.c: calls_0: Subscribing to 213142 [Aug 18 10:34:19] DEBUG[14514] stasis/app.c: Channel '213142' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14656] channel.c: Channel Recorder/ARI-00000036;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[14658] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:19] DEBUG[14658] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:19] DEBUG[14656] channel.c: Channel SIP/zvonobot-00000029 setting write format path: slin -> ulaw [Aug 18 10:34:19] DEBUG[14514] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14514] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14658] http.c: Match made with [ari] [Aug 18 10:34:19] DEBUG[14416] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is joining simple_bridge technology [Aug 18 10:34:19] DEBUG[14416] channel.c: Channel Recorder/ARI-00000024;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Outgoing Call for 79821116898 [Aug 18 10:34:19] DEBUG[14656] channel.c: Channel SIP/zvonobot-00000029 setting read format path: ulaw -> slin [Aug 18 10:34:19] DEBUG[14416] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[14656] channel.c: Channel Recorder/ARI-00000036;2 setting write format path: slin -> slin [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Finding handler for bridges [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology start [Aug 18 10:34:19] DEBUG[14653] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: deferring softmix technology destructor [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:19] DEBUG[14657] app.c: play_and_record: , /var/spool/asterisk/recording/212996_tETNWgXOdyOJtmtuSqFlpvUqMEjxiyua, 'wav' [Aug 18 10:34:19] DEBUG[14513] channel.c: Channel 0x7f0c2c09fc70 'SIP/zvonobot-000000b0' allocated [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:19] DEBUG[14513] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:19] DEBUG[14513] res_stasis.c: calls_0: Subscribing to 213138 [Aug 18 10:34:19] DEBUG[14513] stasis/app.c: Channel '213138' is 1 interested in calls_0 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Outgoing Call for 79821116902 [Aug 18 10:34:19] DEBUG[14655] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:19] VERBOSE[14653] bridge_channel.c: Channel Recorder/ARI-00000035;2 joined 'simple_bridge' stasis-bridge <21515bb0-91f2-4ad5-852f-8721c870cad7> [Aug 18 10:34:19] DEBUG[14416] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: queueing action type:13 sub:1000 [Aug 18 10:34:19] VERBOSE[14655] bridge_channel.c: Channel Announcer/ARI-00000037;2 joined 'simple_bridge' stasis-bridge <0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3> [Aug 18 10:34:19] DEBUG[14513] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:19] DEBUG[14513] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 From: ;tag=as4e77dae5 To: ;tag=as54b5330a Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 309728419 309728419 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16904 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4e77dae5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as54b5330a [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 309728419 309728419 IN IP4 178.62.121.41 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16904 RTP/AVP 0 8 101 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:19] DEBUG[14420] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: stopping mixing thread [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:19] DEBUG[20534] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:19] DEBUG[20534] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: Waiting for mixing thread to die. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:19] DEBUG[14658] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] DEBUG[14658] stasis.c: Creating topic. name: bridge:9fd06d58-bb46-42dd-bd63-c898ede7e980, detail: [Aug 18 10:34:19] DEBUG[14657] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:19] DEBUG[13852] channel.c: Recorder/ARI-00000024;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:19] VERBOSE[14659] chan_sip.c: Audio is at 17048 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 (Checking To) --From tag as4e77dae5 --To-tag as54b5330a [Aug 18 10:34:19] DEBUG[13704] channel.c: SIP/zvonobot-00000013: Dropping redundant connected line update "" <>. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:19] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTCP got report of 100 bytes from 178.62.121.41:18113 [Aug 18 10:34:19] VERBOSE[14657] app.c: x=0, open writing: /var/spool/asterisk/recording/212996_tETNWgXOdyOJtmtuSqFlpvUqMEjxiyua format: wav, 0x7f0c3011a4f0 [Aug 18 10:34:19] DEBUG[14416] channel.c: Channel 0x7f0c9c0ab760 'Announcer/ARI-0000002f;2' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:19] VERBOSE[14660] chan_sip.c: Audio is at 14208 [Aug 18 10:34:19] VERBOSE[14660] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[14660] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] VERBOSE[14660] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Initializing initreq for method INVITE - callid 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116902@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 3 [ 52]: From: ;tag=as563f7715 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 6 [ 60]: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14660] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 849904962 849904962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #139 [Aug 18 10:34:19] DEBUG[14660] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Stopping retransmission on '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:19] VERBOSE[14659] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:19] VERBOSE[14659] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:19] VERBOSE[14659] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:19] DEBUG[14658] stasis.c: Topic 'bridge:9fd06d58-bb46-42dd-bd63-c898ede7e980': 0x7f0c3c078290 created [Aug 18 10:34:19] DEBUG[14658] stasis.c: Creating topic. name: cache:504/bridge:9fd06d58-bb46-42dd-bd63-c898ede7e980, detail: [Aug 18 10:34:19] DEBUG[14403] res_stasis_recording.c: 1629282853.346: Sending record(213006_JPrwZVkIlHRZZBNMHYoLeMMJKYBGrECr.wav) command [Aug 18 10:34:19] DEBUG[14653] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7'. Checking compatability for channels 'SIP/zvonobot-0000000a' and 'Recorder/ARI-00000035;2' [Aug 18 10:34:19] DEBUG[14653] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' can not use native RTP bridge as could not get details [Aug 18 10:34:19] DEBUG[14653] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:19] DEBUG[14653] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14653] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14653] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:19] DEBUG[14653] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7 is already using the new technology. [Aug 18 10:34:19] DEBUG[14653] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: 0x7f0c3c013500(Recorder/ARI-00000035;2) is joining simple_bridge technology [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Got SDP version 309728419 and unique parts [root 309728419 IN IP4 178.62.121.41] [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 309728419 309728419 IN IP4 178.62.121.41... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c059c40) ICE set role failed; no ice instance [Aug 18 10:34:19] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c059c40) RTCP setting address on RTP instance [Aug 18 10:34:19] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c1c05b380 -- Strict RTP learning after remote address set to: 178.62.121.41:16904 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:16904 [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb000c428) from 0x7f0c147e2330 to 0x7f0c1c059e18 [Aug 18 10:34:19] DEBUG[14403] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Initializing initreq for method INVITE - callid 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14653] channel.c: Channel Recorder/ARI-00000035;2 setting read format path: slin -> slin [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0109918) from 0x7f0c147e2330 to 0x7f0c1c059e18 [Aug 18 10:34:19] DEBUG[14403] http.c: HTTP closing session. Top level [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:19] DEBUG[14653] channel.c: Channel SIP/zvonobot-0000000a setting write format path: slin -> ulaw [Aug 18 10:34:19] DEBUG[14653] channel.c: Channel SIP/zvonobot-0000000a setting read format path: ulaw -> slin [Aug 18 10:34:19] DEBUG[14653] channel.c: Channel Recorder/ARI-00000035;2 setting write format path: slin -> slin [Aug 18 10:34:19] WARNING[14331] app.c: No audio available on Recorder/ARI-00000030;1?? [Aug 18 10:34:19] VERBOSE[14331] app.c: User hung up [Aug 18 10:34:19] DEBUG[14331] res_stasis_recording.c: 1629282848.273: Recording complete [Aug 18 10:34:19] DEBUG[14331] channel.c: Channel 0x7f0c180999a0 'Recorder/ARI-00000030;1' hanging up. Refs: 2 [Aug 18 10:34:19] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb006f0b8) from 0x7f0c147e2330 to 0x7f0c1c059e18 [Aug 18 10:34:19] DEBUG[14655] bridge.c: Chose bridge technology softmix [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c059c40) RTCP ignoring duplicate property [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:19] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003c setting read format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003c setting write format path: alaw -> alaw [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c059c40) DTLS - ast_rtp_activate rtp=0x7f0c1c05b380 - setup and perform DTLS' [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c05b380) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:19] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c05b380) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:19] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Strict routing enforced for session 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:19] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:19] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:19] DEBUG[14662] http.c: HTTP opening session. Top level [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116898@178.62.121.41 SIP/2.0 [Aug 18 10:34:19] VERBOSE[14660] dial.c: Called zvonobot/79821116902 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:19] DEBUG[14662] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:19] VERBOSE[14655] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: switching from simple_bridge technology to softmix [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117013@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6f65e568 Max-Forwards: 70 From: ;tag=as4e77dae5 To: ;tag=as54b5330a Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (2) INVITE - 5 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116905@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f Max-Forwards: 70 From: ;tag=as056b2e4e To: Contact: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1088879038 1088879038 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Session timer started: 143 - 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 1768000ms [Aug 18 10:34:19] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:19] DEBUG[14658] stasis.c: Topic 'cache:504/bridge:9fd06d58-bb46-42dd-bd63-c898ede7e980': 0x7f0c3c078070 created [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 3 [ 52]: From: ;tag=as35c0c7eb [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 6 [ 60]: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:19 GMT [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:19] VERBOSE[14659] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116898@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 Max-Forwards: 70 From: ;tag=as35c0c7eb To: Contact: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 144689560 144689560 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #112 [Aug 18 10:34:19] DEBUG[14659] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:19] VERBOSE[14659] dial.c: Called zvonobot/79821116898 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f;received=159.65.48.104 From: ;tag=as42198afd To: ;tag=as3daa4fa6 Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47425851" Content-Length: 0 <-------------> [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f;received=159.65.48.104 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42198afd [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3daa4fa6 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47425851" [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:19] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 (Checking To) --From tag as42198afd --To-tag as3daa4fa6 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:19] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: calling softmix technology constructor [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c9c04b7a0(SIP/zvonobot-0000002f) to dummy bridge temporarily [Aug 18 10:34:19] DEBUG[14661] app.c: play_and_record: , /var/spool/asterisk/recording/213006_JPrwZVkIlHRZZBNMHYoLeMMJKYBGrECr, 'wav' [Aug 18 10:34:19] VERBOSE[13410] dial.c: SIP/zvonobot-0000003c answered [Aug 18 10:34:19] VERBOSE[13410] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003c [Aug 18 10:34:19] DEBUG[13410] stasis/app.c: Channel '213027' is 2 interested in calls_0 [Aug 18 10:34:19] DEBUG[14655] bridge.c: Bridge 0720fe7d-0a53-48b9-97c1-d1fcdba6bfa3: moving 0x7f0c20086d10(Recorder/ARI-00000023;2) to dummy bridge temporarily [Aug 18 10:34:19] WARNING[14655] logger: Log queue threshold (1000) exceeded. Discarding new messages. [Aug 18 10:34:20] WARNING[20531] logger: Logging resumed. 834 messages discarded. [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Stopping retransmission on '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Strict routing enforced for session 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14688] http.c: HTTP opening session. Top level [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:20] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:20] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117013@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b36ecfe Max-Forwards: 70 From: ;tag=as4e77dae5 To: ;tag=as54b5330a Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #144 (3) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #144)) [Aug 18 10:34:20] DEBUG[14539] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:20] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[14539] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[12986] res_rtp_asterisk.c: (0x7f0c2c01b720) RTP 0x7f0c2c01ce60 -- Received packet from 178.62.121.41:18324, dropping due to strict RTP protection. [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116900@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 Max-Forwards: 70 From: ;tag=as154d6f9d To: Contact: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2080919150 2080919150 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14297] chan_sip.c: Hangup call SIP/zvonobot-00000086, SIP callid 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14688] http.c: HTTP Request URI is /ari/channels/213154?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116886&callerId=74950493843 [Aug 18 10:34:20] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[14297] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14297] res_rtp_asterisk.c: (0x7f0c8c03ed40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14297] res_rtp_asterisk.c: (0x7f0c8c03ed40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14297] channel.c: Channel 0x7f0c8c10a180 'SIP/zvonobot-00000086' destroying [Aug 18 10:34:20] DEBUG[14688] http.c: match request [ari/channels/213154] with handler [httpstatus] len 10 [Aug 18 10:34:20] VERBOSE[14683] bridge_channel.c: Channel SIP/zvonobot-0000003c joined 'simple_bridge' stasis-bridge <6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb> [Aug 18 10:34:20] DEBUG[14688] http.c: match request [ari/channels/213154] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (3) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116899@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 Max-Forwards: 70 From: ;tag=as6a49d994 To: Contact: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 370970940 370970940 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14548] res_stasis_playback.c: 1629282851.313: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:20] DEBUG[14548] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:20] DEBUG[14688] http.c: match request [ari/channels/213154] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213103': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213103' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14548] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14297] stasis.c: Destroying topic. name: cache:326/channel:213103, detail: [Aug 18 10:34:20] DEBUG[14297] stasis.c: Topic 'cache:326/channel:213103': 0x7f0c8c04f7e0 destroyed [Aug 18 10:34:20] DEBUG[14297] stasis.c: Destroying topic. name: channel:213103, detail: [Aug 18 10:34:20] DEBUG[14297] stasis.c: Topic 'channel:213103': 0x7f0c8c07a110 destroyed [Aug 18 10:34:20] DEBUG[14685] bridge_channel.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03: pushing 0x7f0c8c05eba0(SIP/zvonobot-0000004c) [Aug 18 10:34:20] VERBOSE[14685] bridge_channel.c: Channel SIP/zvonobot-0000004c joined 'simple_bridge' stasis-bridge <804476f3-9df1-4495-8c76-b406f7162d03> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (2) INVITE - 5 [Aug 18 10:34:20] DEBUG[14689] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 849904962 849904962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14690] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14692] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14685] bridge_native_rtp.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03' can not use native RTP bridge as two channels are required [Aug 18 10:34:20] DEBUG[14688] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14689] http.c: HTTP Request URI is /ari/playbacks/0f587303-a3ba-4e54-a2d8-50f35c6527df [Aug 18 10:34:20] DEBUG[14689] http.c: match request [ari/playbacks/0f587303-a3ba-4e54-a2d8-50f35c6527df] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14689] http.c: match request [ari/playbacks/0f587303-a3ba-4e54-a2d8-50f35c6527df] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14689] http.c: match request [ari/playbacks/0f587303-a3ba-4e54-a2d8-50f35c6527df] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14689] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14692] http.c: HTTP Request URI is /ari/channels/213156?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116884&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Finding handler for playbacks/0f587303-a3ba-4e54-a2d8-50f35c6527df [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Finding handler for playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Finding handler for 0f587303-a3ba-4e54-a2d8-50f35c6527df [Aug 18 10:34:20] DEBUG[14689] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14689] res_ari.c: No explicit handler found for 0f587303-a3ba-4e54-a2d8-50f35c6527df. Using wildcard playbackId. [Aug 18 10:34:20] DEBUG[14690] http.c: HTTP Request URI is /ari/channels/213103 [Aug 18 10:34:20] DEBUG[14448] channel.c: Soft-Hanging (0x20) up channel 'Snoop/212964-00000000' [Aug 18 10:34:20] DEBUG[14448] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: cache:513/channel:1629282860.444, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:513/channel:1629282860.444': 0x7f0c30109130 destroyed [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282860.444, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.444': 0x7f0c300fba90 destroyed [Aug 18 10:34:20] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:11', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000085', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213095', '')] [Aug 18 10:34:20] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000092 - start 1629282853.344161 answer 0.000000 end 1629282859.951419 dur 6.607 bill 1629282859.951 dispo NO ANSWER [Aug 18 10:34:20] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000093 - start 1629282853.383601 answer 0.000000 end 1629282859.966486 dur 6.582 bill 1629282859.966 dispo NO ANSWER [Aug 18 10:34:20] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000090 - start 1629282853.279220 answer 0.000000 end 1629282859.968955 dur 6.689 bill 1629282859.968 dispo NO ANSWER [Aug 18 10:34:20] DEBUG[14695] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14695] http.c: HTTP Request URI is /ari/channels/robot_213007 [Aug 18 10:34:20] DEBUG[14448] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14685] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:20] DEBUG[14692] http.c: match request [ari/channels/213156] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14696] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14688] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14695] http.c: match request [ari/channels/robot_213007] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14695] http.c: match request [ari/channels/robot_213007] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14692] http.c: match request [ari/channels/213156] with handler [phoneprov] len 9 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652 Max-Forwards: 70 From: ;tag=as4e94a883 To: ;tag=as181bb145 Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6ccf4cd6", response="3b85051dbb1a77b0db3c978d5568960d" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as4e94a883 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as181bb145 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="6ccf4cd6", response="3b85051dbb1a77b0db3c978d5568960d" [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 (Checking From) --From tag as4e94a883 --To-tag as181bb145 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:20] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14695] http.c: match request [ari/channels/robot_213007] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14685] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[14692] http.c: match request [ari/channels/213156] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14697] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1e5eb652;received=178.62.121.41 From: ;tag=as4e94a883 To: ;tag=as181bb145 Call-ID: 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14690] http.c: match request [ari/channels/213103] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14695] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #129 (1) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #129)) [Aug 18 10:34:20] DEBUG[14692] http.c: Match made with [ari] [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116897@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9 Max-Forwards: 70 From: ;tag=as5acf84f3 To: Contact: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168069357 1168069357 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18198 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14689] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #7 (1) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #7)) [Aug 18 10:34:20] DEBUG[14696] http.c: HTTP Request URI is /ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:34:20] DEBUG[14692] http.c: HTTP consuming request body [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116901@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b Max-Forwards: 70 From: ;tag=as3a1d6e7b To: Contact: Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1147590624 1147590624 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (2) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116898@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 Max-Forwards: 70 From: ;tag=as35c0c7eb To: Contact: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 144689560 144689560 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c03ed40) DTLS stop [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c03ed40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c03ed40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c03ed40) ICE RTP transport deallocating [Aug 18 10:34:20] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8c03ed40' [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734;received=159.65.48.104 From: ;tag=as611ff9f7 To: ;tag=as0807854b Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51d63101" Content-Length: 0 <-------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3f41e734;received=159.65.48.104 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as611ff9f7 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0807854b [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51d63101" [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 (Checking To) --From tag as611ff9f7 --To-tag as0807854b [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3abab2ad5de75687001dc3381ae29b62@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (3) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116905@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f Max-Forwards: 70 From: ;tag=as056b2e4e To: Contact: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1088879038 1088879038 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14685] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Finding handler for channels/213156 [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Finding handler for 213156 [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking channels create: Didn't match 213156 [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14692] res_ari.c: Checking channels externalMedia: Didn't match 213156 [Aug 18 10:34:20] DEBUG[14692] res_ari.c: No explicit handler found for 213156. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14697] http.c: HTTP Request URI is /ari/channels/213157?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116883&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14689] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14696] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14690] http.c: match request [ari/channels/213103] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Finding handler for channels/robot_213007 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Finding handler for channels/213154 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14696] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14696] http.c: match request [ari/bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14690] http.c: match request [ari/channels/213103] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14702] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14690] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14697] http.c: match request [ari/channels/213157] with handler [httpstatus] len 10 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b;received=159.65.48.104 From: ;tag=as17300792 To: ;tag=as6b1c1c07 Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c99099" Content-Length: 0 <-------------> [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14685] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Finding handler for channels/213103 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK691b360b;received=159.65.48.104 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as17300792 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6b1c1c07 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:20] DEBUG[14696] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[14702] http.c: HTTP Request URI is /ari/channels/213155?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116885&callerId=74950493843 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Finding handler for bridges/7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Finding handler for bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Finding handler for 7421ba4f-6229-4eeb-b806-91ebc84ff38c [Aug 18 10:34:20] DEBUG[14696] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14685] bridge.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03 is already using the new technology. [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:20] DEBUG[14697] http.c: match request [ari/channels/213157] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Finding handler for 213154 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking channels create: Didn't match 213154 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14688] res_ari.c: Checking channels externalMedia: Didn't match 213154 [Aug 18 10:34:20] DEBUG[14688] res_ari.c: No explicit handler found for 213154. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14696] res_ari.c: No explicit handler found for 7421ba4f-6229-4eeb-b806-91ebc84ff38c. Using wildcard bridgeId. [Aug 18 10:34:20] DEBUG[14685] bridge.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03: 0x7f0c8c05eba0(SIP/zvonobot-0000004c) is joining simple_bridge technology [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c99099" [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 (Checking To) --From tag as17300792 --To-tag as6b1c1c07 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1df6c2155486adc004a0a603618a3f64@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (5) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1997886737 1997886737 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 From: ;tag=as64e6e544 To: ;tag=as189a4383 Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" Content-Length: 0 <-------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as64e6e544 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as189a4383 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: = Looking for Call ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 (Checking To) --From tag as64e6e544 --To-tag as189a4383 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:20] DEBUG[14702] http.c: match request [ari/channels/213155] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Finding handler for 213103 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 Max-Forwards: 70 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as20bcc5bd [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as31e40966 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking From) --From tag as20bcc5bd --To-tag as31e40966 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:20] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14696] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: telling all channels to leave the party [Aug 18 10:34:20] DEBUG[14696] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:20] DEBUG[14696] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: queueing action type:13 sub:1001 [Aug 18 10:34:20] DEBUG[14696] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282860.445, detail: [Aug 18 10:34:20] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14702] http.c: match request [ari/channels/213155] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14702] http.c: match request [ari/channels/213155] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14702] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14702] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14705] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling stasis bridge destructor [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology stop [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 7421ba4f-6229-4eeb-b806-91ebc84ff38c: calling simple_bridge technology destructor [Aug 18 10:34:20] DEBUG[14706] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking channels create: Didn't match 213103 [Aug 18 10:34:20] DEBUG[14706] http.c: HTTP Request URI is /ari/channels/213159?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116881&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14706] http.c: match request [ari/channels/213159] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14705] http.c: HTTP Request URI is /ari/channels/213158?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116882&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14696] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14690] res_ari.c: Checking channels externalMedia: Didn't match 213103 [Aug 18 10:34:20] DEBUG[14690] res_ari.c: No explicit handler found for 213103. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14706] http.c: match request [ari/channels/213159] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14706] http.c: match request [ari/channels/213159] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14683] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb' can not use native RTP bridge as two channels are required [Aug 18 10:34:20] DEBUG[14706] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14707] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:20] DEBUG[14697] http.c: match request [ari/channels/213157] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.445': 0x7f0c3006ade0 created [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: cache:514/channel:1629282860.445, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:514/channel:1629282860.445': 0x7f0c30070390 created [Aug 18 10:34:20] DEBUG[14706] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14705] http.c: match request [ari/channels/213158] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Finding handler for robot_213007 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: bridge '7421ba4f-6229-4eeb-b806-91ebc84ff38c' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Finding handler for channels/213159 [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14708] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking channels create: Didn't match robot_213007 [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: cache:37/bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c, detail: [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Finding handler for 213159 [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking channels create: Didn't match 213159 [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14708] http.c: HTTP Request URI is /ari/channels/213162?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116878&callerId=74950493843 [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'cache:37/bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c': 0x7f0c9c002110 destroyed [Aug 18 10:34:20] DEBUG[14706] res_ari.c: Checking channels externalMedia: Didn't match 213159 [Aug 18 10:34:20] DEBUG[14708] http.c: match request [ari/channels/213162] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14708] http.c: match request [ari/channels/213162] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14708] http.c: match request [ari/channels/213162] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14706] res_ari.c: No explicit handler found for 213159. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14695] res_ari.c: Checking channels externalMedia: Didn't match robot_213007 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Finding handler for channels/213155 [Aug 18 10:34:20] DEBUG[14705] http.c: match request [ari/channels/213158] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14707] http.c: HTTP Request URI is /ari/bridges/87d87304-31e6-4326-b367-680423189269 [Aug 18 10:34:20] DEBUG[14697] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14708] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c, detail: [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'bridge:7421ba4f-6229-4eeb-b806-91ebc84ff38c': 0x7f0c9c0130d0 destroyed [Aug 18 10:34:20] DEBUG[14708] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14685] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTP changing ssrc from 484434086 to 108347490 due to a source change [Aug 18 10:34:20] DEBUG[13563] stasis/app.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03' is 2 interested in calls_0 [Aug 18 10:34:20] DEBUG[14682] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[14683] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:20] DEBUG[14682] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14710] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14695] res_ari.c: No explicit handler found for robot_213007. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14683] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[14705] http.c: match request [ari/channels/213158] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14709] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14710] http.c: HTTP Request URI is /ari/channels/213161?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116879&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: Allocating new SIP dialog for 61a3a4197754281555a3ec9546260e83@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:20] DEBUG[14688] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c18009d30' [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) RTP allocated port 14616 [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE creating session 0.0.0.0:14616 (14616) [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE create [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE add system candidates [Aug 18 10:34:20] DEBUG[14688] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:20] DEBUG[14688] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE add candidate: 159.65.48.104:14616, 2130706431 [Aug 18 10:34:20] DEBUG[14688] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:20] DEBUG[14688] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: cache:514/channel:1629282860.445, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:514/channel:1629282860.445': 0x7f0c30070390 destroyed [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282860.445, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.445': 0x7f0c3006ade0 destroyed [Aug 18 10:34:20] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:11', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000086', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213103', '')] [Aug 18 10:34:20] DEBUG[14705] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14705] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Finding handler for channels/213158 [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Finding handler for 213158 [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking channels create: Didn't match 213158 [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14705] res_ari.c: Checking channels externalMedia: Didn't match 213158 [Aug 18 10:34:20] DEBUG[14705] res_ari.c: No explicit handler found for 213158. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14709] http.c: HTTP Request URI is /ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/record?name=213041_SDPjzwBTGJBwjIEyyHVbOCOACYofXdxs&format=wav [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Finding handler for channels/213162 [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14707] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14683] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[14710] http.c: match request [ari/channels/213161] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14709] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/record] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Finding handler for 213155 [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14709] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/record] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking channels create: Didn't match 213155 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14710] http.c: match request [ari/channels/213161] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE add candidate: 10.131.0.10:14616, 2130706431 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: Checking channels externalMedia: Didn't match 213155 [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14709] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/record] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14702] res_ari.c: No explicit handler found for 213155. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14707] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14688] rtp_engine.c: RTP instance '0x7f0c18009d30' is setup and ready to go [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) ICE stopped [Aug 18 10:34:20] DEBUG[14688] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:20] DEBUG[14688] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:20] DEBUG[14688] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:20] DEBUG[14688] res_rtp_asterisk.c: (0x7f0c18009d30) RTCP setup on RTP instance [Aug 18 10:34:20] VERBOSE[14688] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:20] DEBUG[14688] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14688] chan_sip.c: SIP call-id changed from '61a3a4197754281555a3ec9546260e83@127.0.1.1:5060' to '1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060' [Aug 18 10:34:20] DEBUG[14709] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14711] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14710] http.c: match request [ari/channels/213161] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Finding handler for bridges/804476f3-9df1-4495-8c76-b406f7162d03/record [Aug 18 10:34:20] DEBUG[14697] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Finding handler for 213162 [Aug 18 10:34:20] DEBUG[14710] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking channels create: Didn't match 213162 [Aug 18 10:34:20] DEBUG[14711] http.c: HTTP Request URI is /ari/channels/213160?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116880&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14683] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Finding handler for bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Finding handler for 804476f3-9df1-4495-8c76-b406f7162d03 [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14709] res_ari.c: No explicit handler found for 804476f3-9df1-4495-8c76-b406f7162d03. Using wildcard bridgeId. [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Finding handler for record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:20] DEBUG[14709] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:20] DEBUG[14709] stasis.c: Creating topic. name: channel:1629282860.447, detail: [Aug 18 10:34:20] DEBUG[14709] stasis.c: Topic 'channel:1629282860.447': 0x7f0c7c011720 created [Aug 18 10:34:20] DEBUG[14709] stasis.c: Creating topic. name: cache:515/channel:1629282860.447, detail: [Aug 18 10:34:20] DEBUG[14709] stasis.c: Topic 'cache:515/channel:1629282860.447': 0x7f0c7c0121a0 created [Aug 18 10:34:20] DEBUG[14708] res_ari.c: Checking channels externalMedia: Didn't match 213162 [Aug 18 10:34:20] DEBUG[14708] res_ari.c: No explicit handler found for 213162. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14710] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14683] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb is already using the new technology. [Aug 18 10:34:20] DEBUG[14688] stasis.c: Creating topic. name: channel:213154, detail: [Aug 18 10:34:20] DEBUG[14711] http.c: match request [ari/channels/213160] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Finding handler for channels/213161 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14707] http.c: match request [ari/bridges/87d87304-31e6-4326-b367-680423189269] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14711] http.c: match request [ari/channels/213160] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14688] stasis.c: Topic 'channel:213154': 0x7f0c1802ba70 created [Aug 18 10:34:20] DEBUG[14683] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: 0x7f0c70099da0(SIP/zvonobot-0000003c) is joining simple_bridge technology [Aug 18 10:34:20] DEBUG[14712] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14688] stasis.c: Creating topic. name: cache:516/channel:213154, detail: [Aug 18 10:34:20] DEBUG[14707] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14711] http.c: match request [ari/channels/213160] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14712] http.c: HTTP Request URI is /ari/channels/213163?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116877&callerId=74950493843 [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] DEBUG[14683] res_rtp_asterisk.c: (0x7f0c1c059c40) RTP changing ssrc from 2083607692 to 430102768 due to a source change [Aug 18 10:34:20] DEBUG[14712] http.c: match request [ari/channels/213163] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14712] http.c: match request [ari/channels/213163] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14688] stasis.c: Topic 'cache:516/channel:213154': 0x7f0c1802c4f0 created [Aug 18 10:34:20] DEBUG[14713] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14713] http.c: HTTP Request URI is /ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/record?name=213027_LzhbtVqcAVRxNKmskAtSCBZXSwnXqDKu&format=wav [Aug 18 10:34:20] DEBUG[14713] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/record] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14713] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/record] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14713] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/record] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14713] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Finding handler for bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/record [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Finding handler for bridges [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[13410] stasis/app.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb' is 2 interested in calls_0 [Aug 18 10:34:20] DEBUG[14711] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14675] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Finding handler for 213161 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking channels create: Didn't match 213161 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14710] res_ari.c: Checking channels externalMedia: Didn't match 213161 [Aug 18 10:34:20] DEBUG[14710] res_ari.c: No explicit handler found for 213161. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14675] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14712] http.c: match request [ari/channels/213163] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Finding handler for channels/213157 [Aug 18 10:34:20] DEBUG[14711] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14712] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Finding handler for channels/213160 [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14712] http.c: HTTP consuming request body [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Finding handler for 213160 [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Finding handler for channels/213163 [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Finding handler for bridges/87d87304-31e6-4326-b367-680423189269 [Aug 18 10:34:20] DEBUG[14321] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:20] DEBUG[14321] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking channels create: Didn't match 213160 [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Finding handler for 213157 [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking channels create: Didn't match 213157 [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14697] res_ari.c: Checking channels externalMedia: Didn't match 213157 [Aug 18 10:34:20] DEBUG[14697] res_ari.c: No explicit handler found for 213157. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Finding handler for bridges [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14711] res_ari.c: Checking channels externalMedia: Didn't match 213160 [Aug 18 10:34:20] DEBUG[14711] res_ari.c: No explicit handler found for 213160. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Finding handler for 213163 [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:20] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking channels create: Didn't match 213163 [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Finding handler for 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: Allocating new SIP dialog for 4bdcb5a20471d82814de420b3d4806a2@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:20] DEBUG[14706] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c400bd8b0' [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) RTP allocated port 11948 [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE creating session 0.0.0.0:11948 (11948) [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE create [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14713] res_ari.c: No explicit handler found for 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb. Using wildcard bridgeId. [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:20] DEBUG[14712] res_ari.c: Checking channels externalMedia: Didn't match 213163 [Aug 18 10:34:20] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE add system candidates [Aug 18 10:34:20] DEBUG[14712] res_ari.c: No explicit handler found for 213163. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Finding handler for record [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09;received=178.62.121.41 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: Allocating new SIP dialog for 0c34a1e010e388a561dcf0f45bc608f5@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:20] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: Allocating new SIP dialog for 4bd727d46923c81b35ccac3a0bd5ea41@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:20] DEBUG[14708] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7012f9f0' [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) RTP allocated port 11726 [Aug 18 10:34:20] DEBUG[14706] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:20] DEBUG[14706] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE add candidate: 159.65.48.104:11948, 2130706431 [Aug 18 10:34:20] DEBUG[14706] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:20] DEBUG[14706] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE add candidate: 10.131.0.10:11948, 2130706431 [Aug 18 10:34:20] DEBUG[14706] rtp_engine.c: RTP instance '0x7f0c400bd8b0' is setup and ready to go [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE stopped [Aug 18 10:34:20] DEBUG[14706] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:20] DEBUG[14706] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:20] DEBUG[14706] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:20] DEBUG[14706] res_rtp_asterisk.c: (0x7f0c400bd8b0) RTCP setup on RTP instance [Aug 18 10:34:20] VERBOSE[14706] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE creating session 0.0.0.0:11726 (11726) [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE create [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE add system candidates [Aug 18 10:34:20] DEBUG[14708] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:20] DEBUG[14708] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (3) INVITE - 5 [Aug 18 10:34:20] DEBUG[14702] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c15a110' [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) RTP allocated port 16374 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE add candidate: 159.65.48.104:11726, 2130706431 [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE creating session 0.0.0.0:16374 (16374) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116904@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face Max-Forwards: 70 From: ;tag=as1a5706e7 To: Contact: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 641212001 641212001 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13764 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE create [Aug 18 10:34:20] DEBUG[14307] chan_sip.c: Hangup call SIP/zvonobot-00000088, SIP callid 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14307] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:20] DEBUG[14307] res_rtp_asterisk.c: (0x7f0c940b4340) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14307] res_rtp_asterisk.c: (0x7f0c940b4340) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14307] channel.c: Channel 0x7f0c940bd420 'SIP/zvonobot-00000088' destroying [Aug 18 10:34:20] DEBUG[12968] chan_sip.c: Hangup call SIP/zvonobot-00000012, SIP callid 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14216] channel.c: Channel 0x7f0c2412f860 'Announcer/ARI-00000038;2' allocated [Aug 18 10:34:20] DEBUG[14303] chan_sip.c: Hangup call SIP/zvonobot-00000087, SIP callid 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14318] chan_sip.c: Hangup call SIP/zvonobot-0000008b, SIP callid 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14706] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:20] DEBUG[14318] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:20] DEBUG[14318] res_rtp_asterisk.c: (0x7f0c840953a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[13059] channel.c: Channel 0x7f0ca4006700 'Recorder/ARI-00000000;1' destroying [Aug 18 10:34:20] DEBUG[13059] stasis.c: Destroying topic. name: cache:38/channel:1629282827.30, detail: [Aug 18 10:34:20] DEBUG[13059] stasis.c: Topic 'cache:38/channel:1629282827.30': 0x7f0ca40089a0 destroyed [Aug 18 10:34:20] DEBUG[13059] stasis.c: Destroying topic. name: channel:1629282827.30, detail: [Aug 18 10:34:20] DEBUG[13059] stasis.c: Topic 'channel:1629282827.30': 0x7f0ca4008c20 destroyed [Aug 18 10:34:20] DEBUG[14216] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:20] DEBUG[14216] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000038;1' [Aug 18 10:34:20] DEBUG[14349] channel.c: Channel 0x7f0c80025890 'Recorder/ARI-0000002e;1' destroying [Aug 18 10:34:20] DEBUG[14349] stasis.c: Destroying topic. name: cache:312/channel:1629282847.267, detail: [Aug 18 10:34:20] DEBUG[14349] stasis.c: Topic 'cache:312/channel:1629282847.267': 0x7f0c80030150 destroyed [Aug 18 10:34:20] DEBUG[14349] stasis.c: Destroying topic. name: channel:1629282847.267, detail: [Aug 18 10:34:20] DEBUG[14349] stasis.c: Topic 'channel:1629282847.267': 0x7f0c8002b2b0 destroyed [Aug 18 10:34:20] DEBUG[12968] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[12968] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14450] channel.c: Channel 0x7f0c200b0230 'Recorder/ARI-00000039;2' allocated [Aug 18 10:34:20] DEBUG[14303] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:20] DEBUG[14303] res_rtp_asterisk.c: (0x7f0c3808f7a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14303] res_rtp_asterisk.c: (0x7f0c3808f7a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14303] channel.c: Channel 0x7f0c38082e90 'SIP/zvonobot-00000087' destroying [Aug 18 10:34:20] DEBUG[14308] chan_sip.c: Hangup call SIP/zvonobot-00000089, SIP callid 7f9397da0c066f606bb6d6845209f912@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14308] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:20] DEBUG[14308] res_rtp_asterisk.c: (0x7f0c900af9f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14308] res_rtp_asterisk.c: (0x7f0c900af9f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14308] channel.c: Channel 0x7f0c900b6f50 'SIP/zvonobot-00000089' destroying [Aug 18 10:34:20] DEBUG[14318] res_rtp_asterisk.c: (0x7f0c840953a0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:20] DEBUG[14451] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:20] DEBUG[13226] bridge_channel.c: Setting 0x7f0c2000f4c0(Snoop/212983-00000001) state from:0 to:1 [Aug 18 10:34:20] DEBUG[13226] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: pulling 0x7f0c2000f4c0(Snoop/212983-00000001) [Aug 18 10:34:20] VERBOSE[13226] bridge_channel.c: Channel Snoop/212983-00000001 left 'simple_bridge' stasis-bridge <3f704757-87e2-45e5-8aa9-92ed6ea9feee> [Aug 18 10:34:20] DEBUG[13226] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c2000f4c0(Snoop/212983-00000001) is leaving simple_bridge technology [Aug 18 10:34:20] DEBUG[13226] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' can not use native RTP bridge as two channels are required [Aug 18 10:34:20] DEBUG[13226] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:20] DEBUG[13226] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[13226] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[13226] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:20] DEBUG[13226] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee is already using the new technology. [Aug 18 10:34:20] DEBUG[13226] bridge_channel.c: Bridge is returning 0x7f0c2000f4c0(Snoop/212983-00000001) to read format slin [Aug 18 10:34:20] DEBUG[13226] channel.c: Channel Snoop/212983-00000001 setting read format path: slin -> slin [Aug 18 10:34:20] DEBUG[13226] bridge_channel.c: Bridge is returning 0x7f0c2000f4c0(Snoop/212983-00000001) to write format slin [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14310] bridge_channel.c: Setting 0x7f0c80044670(Recorder/ARI-0000002e;2) state from:0 to:1 [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:20] VERBOSE[13211] app.c: User hung up [Aug 18 10:34:20] DEBUG[13211] res_stasis_recording.c: 1629282830.60: Recording complete [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Finding handler for 87d87304-31e6-4326-b367-680423189269 [Aug 18 10:34:20] DEBUG[14707] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14707] res_ari.c: No explicit handler found for 87d87304-31e6-4326-b367-680423189269. Using wildcard bridgeId. [Aug 18 10:34:20] DEBUG[14707] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: telling all channels to leave the party [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213094': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213094' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213101': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213101' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14450] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:20] DEBUG[13226] channel.c: Channel Snoop/212983-00000001 setting write format path: slin -> slin [Aug 18 10:34:20] DEBUG[14308] stasis.c: Destroying topic. name: cache:330/channel:213101, detail: [Aug 18 10:34:20] DEBUG[14308] stasis.c: Topic 'cache:330/channel:213101': 0x7f0c900755c0 destroyed [Aug 18 10:34:20] DEBUG[14308] stasis.c: Destroying topic. name: channel:213101, detail: [Aug 18 10:34:20] DEBUG[14308] stasis.c: Topic 'channel:213101': 0x7f0c900811a0 destroyed [Aug 18 10:34:20] DEBUG[13210] channel.c: Channel 0x7f0c10035e20 'Recorder/ARI-00000003;2' destroying [Aug 18 10:34:20] DEBUG[14451] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14318] channel.c: Channel 0x7f0c8413e2c0 'SIP/zvonobot-0000008b' destroying [Aug 18 10:34:20] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14707] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:20] DEBUG[14707] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: queueing action type:13 sub:1001 [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:20] DEBUG[14303] stasis.c: Destroying topic. name: cache:327/channel:213094, detail: [Aug 18 10:34:20] DEBUG[14303] stasis.c: Topic 'cache:327/channel:213094': 0x7f0c3805e1b0 destroyed [Aug 18 10:34:20] DEBUG[14303] stasis.c: Destroying topic. name: channel:213094, detail: [Aug 18 10:34:20] DEBUG[14303] stasis.c: Topic 'channel:213094': 0x7f0c3804c510 destroyed [Aug 18 10:34:20] DEBUG[14707] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:20] DEBUG[14707] http.c: HTTP closing session. Top level [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE add system candidates [Aug 18 10:34:20] DEBUG[14702] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:20] DEBUG[14702] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE add candidate: 159.65.48.104:16374, 2130706431 [Aug 18 10:34:20] DEBUG[14702] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:20] DEBUG[14702] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE add candidate: 10.131.0.10:16374, 2130706431 [Aug 18 10:34:20] DEBUG[14702] rtp_engine.c: RTP instance '0x7f0c3c15a110' is setup and ready to go [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) ICE stopped [Aug 18 10:34:20] DEBUG[14702] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:20] DEBUG[14702] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:20] DEBUG[14702] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:20] DEBUG[14702] res_rtp_asterisk.c: (0x7f0c3c15a110) RTCP setup on RTP instance [Aug 18 10:34:20] VERBOSE[14702] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:20] DEBUG[14702] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14702] chan_sip.c: SIP call-id changed from '4bd727d46923c81b35ccac3a0bd5ea41@127.0.1.1:5060' to '0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060' [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: Allocating new SIP dialog for 6c8e589c705d70ba778bd36a30d3a0ac@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:20] DEBUG[14710] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7807cb80' [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) RTP allocated port 16950 [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE creating session 0.0.0.0:16950 (16950) [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE create [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE add system candidates [Aug 18 10:34:20] DEBUG[14710] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:20] DEBUG[14710] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE add candidate: 159.65.48.104:16950, 2130706431 [Aug 18 10:34:20] DEBUG[14710] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:20] DEBUG[14710] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE add candidate: 10.131.0.10:16950, 2130706431 [Aug 18 10:34:20] DEBUG[14710] rtp_engine.c: RTP instance '0x7f0c7807cb80' is setup and ready to go [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) ICE stopped [Aug 18 10:34:20] DEBUG[14710] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:20] DEBUG[14710] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:20] DEBUG[14710] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:20] DEBUG[14710] res_rtp_asterisk.c: (0x7f0c7807cb80) RTCP setup on RTP instance [Aug 18 10:34:20] VERBOSE[14710] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:20] DEBUG[14710] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14710] chan_sip.c: SIP call-id changed from '6c8e589c705d70ba778bd36a30d3a0ac@127.0.1.1:5060' to '31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060' [Aug 18 10:34:20] DEBUG[14710] stasis.c: Creating topic. name: channel:213161, detail: [Aug 18 10:34:20] DEBUG[14710] stasis.c: Topic 'channel:213161': 0x7f0c780c3a60 created [Aug 18 10:34:20] DEBUG[14710] stasis.c: Creating topic. name: cache:517/channel:213161, detail: [Aug 18 10:34:20] DEBUG[14710] stasis.c: Topic 'cache:517/channel:213161': 0x7f0c780c44c0 created [Aug 18 10:34:20] DEBUG[14702] stasis.c: Creating topic. name: channel:213155, detail: [Aug 18 10:34:20] DEBUG[14702] stasis.c: Topic 'channel:213155': 0x7f0c3c168f90 created [Aug 18 10:34:20] DEBUG[14702] stasis.c: Creating topic. name: cache:518/channel:213155, detail: [Aug 18 10:34:20] DEBUG[13211] channel.c: Channel 0x7f0c1002e710 'Recorder/ARI-00000003;1' hanging up. Refs: 2 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #77 (5) INVITE - 5 [Aug 18 10:34:20] DEBUG[14717] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14702] stasis.c: Topic 'cache:518/channel:213155': 0x7f0c3c1699c0 created [Aug 18 10:34:20] DEBUG[13226] stasis/control.c: 1629282830.62, 3f704757-87e2-45e5-8aa9-92ed6ea9feee: Channel was departed from bridge [Aug 18 10:34:20] DEBUG[14715] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c24057ae0(Announcer/ARI-00000038;2) is joining [Aug 18 10:34:20] DEBUG[14719] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:20] DEBUG[14717] http.c: HTTP Request URI is /ari/channels/213094 [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14719] http.c: HTTP Request URI is /ari/channels/213102 [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: calling stasis bridge destructor [Aug 18 10:34:20] DEBUG[13318] channel.c: Channel 0x7f0c280925d0 'Announcer/ARI-00000009;1' destroying [Aug 18 10:34:20] DEBUG[13316] channel.c: Channel 0x7f0c28098390 'Announcer/ARI-00000009;2' destroying [Aug 18 10:34:20] DEBUG[13226] stasis/app.c: bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee': is 3 interested in calls_0 [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213102': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[13214] stasis/control.c: 1629282830.62: Channel departing bridge [Aug 18 10:34:20] DEBUG[13214] bridge.c: Waiting for 0x7f0c2000f4c0(Snoop/212983-00000001) bridge thread to die. [Aug 18 10:34:20] DEBUG[13316] stasis.c: Destroying topic. name: cache:105/channel:1629282832.90, detail: [Aug 18 10:34:20] DEBUG[13316] stasis.c: Topic 'cache:105/channel:1629282832.90': 0x7f0c28010e10 destroyed [Aug 18 10:34:20] DEBUG[13316] stasis.c: Destroying topic. name: channel:1629282832.90, detail: [Aug 18 10:34:20] DEBUG[13316] stasis.c: Topic 'channel:1629282832.90': 0x7f0c28010c40 destroyed [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: calling simple_bridge technology stop [Aug 18 10:34:20] DEBUG[14664] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[13226] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213102' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14713] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:20] DEBUG[14716] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0c200663a0(Recorder/ARI-00000039;2) is joining [Aug 18 10:34:20] DEBUG[13387] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:20] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #77)) [Aug 18 10:34:20] DEBUG[13394] res_rtp_asterisk.c: (0x7f0c840529d0) RTP audio difference is 688, ms is 63 [Aug 18 10:34:20] DEBUG[14310] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pulling 0x7f0c80044670(Recorder/ARI-0000002e;2) [Aug 18 10:34:20] DEBUG[20534] bridge.c: Bridge 87d87304-31e6-4326-b367-680423189269: calling simple_bridge technology destructor [Aug 18 10:34:20] VERBOSE[14310] bridge_channel.c: Channel Recorder/ARI-0000002e;2 left 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:20] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14713] stasis.c: Creating topic. name: channel:1629282860.450, detail: [Aug 18 10:34:20] DEBUG[14713] stasis.c: Topic 'channel:1629282860.450': 0x7f0c8c048d20 created [Aug 18 10:34:20] DEBUG[14713] stasis.c: Creating topic. name: cache:519/channel:1629282860.450, detail: [Aug 18 10:34:20] DEBUG[14713] stasis.c: Topic 'cache:519/channel:1629282860.450': 0x7f0c8c055680 created [Aug 18 10:34:20] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:20] DEBUG[14310] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c80044670(Recorder/ARI-0000002e;2) is leaving simple_bridge technology [Aug 18 10:34:20] DEBUG[14719] http.c: match request [ari/channels/213102] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] DEBUG[14708] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:20] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14719] http.c: match request [ari/channels/213102] with handler [phoneprov] len 9 [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 171494189 171494189 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14784 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14719] http.c: match request [ari/channels/213102] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] DEBUG[13214] stasis/app.c: channel '1629282830.62': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[14708] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:20] DEBUG[13214] stasis/app.c: channel '1629282830.62' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP audio difference is 744, ms is 113 [Aug 18 10:34:20] DEBUG[14718] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:20] DEBUG[14719] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:20] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 672, ms is 62 [Aug 18 10:34:20] DEBUG[13214] channel.c: Channel 0x7f0c2402e210 'Snoop/212983-00000001' hanging up. Refs: 3 [Aug 18 10:34:20] DEBUG[14720] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213099': is 0 interested in calls_0 [Aug 18 10:34:20] DEBUG[14307] stasis.c: Destroying topic. name: cache:328/channel:213102, detail: [Aug 18 10:34:20] DEBUG[14307] stasis.c: Topic 'cache:328/channel:213102': 0x7f0c940af6a0 destroyed [Aug 18 10:34:20] DEBUG[14307] stasis.c: Destroying topic. name: channel:213102, detail: [Aug 18 10:34:20] DEBUG[14307] stasis.c: Topic 'channel:213102': 0x7f0c940ae940 destroyed [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: channel '213099' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14310] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:20] DEBUG[14718] http.c: HTTP Request URI is /ari/channels/213101 [Aug 18 10:34:20] DEBUG[14720] http.c: HTTP Request URI is /ari/channels/213099 [Aug 18 10:34:20] DEBUG[14310] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:20] DEBUG[14706] chan_sip.c: SIP call-id changed from '4bdcb5a20471d82814de420b3d4806a2@127.0.1.1:5060' to '2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060' [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE add candidate: 10.131.0.10:11726, 2130706431 [Aug 18 10:34:20] DEBUG[14310] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[14717] http.c: match request [ari/channels/213094] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #129 (2) INVITE - 5 [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Finding handler for channels/213102 [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14310] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282860.452, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.452': 0x7f0c3006dbf0 created [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: cache:520/channel:1629282860.452, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:520/channel:1629282860.452': 0x7f0c3012a2b0 created [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: cache:520/channel:1629282860.452, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:520/channel:1629282860.452': 0x7f0c3012a2b0 destroyed [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282860.452, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.452': 0x7f0c3006dbf0 destroyed [Aug 18 10:34:20] DEBUG[14720] http.c: match request [ari/channels/213099] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[13318] stasis.c: Destroying topic. name: cache:104/channel:1629282832.89, detail: [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14310] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:20] DEBUG[14310] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[13210] stasis.c: Destroying topic. name: cache:71/channel:1629282830.61, detail: [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: cache:331/channel:213099, detail: [Aug 18 10:34:20] DEBUG[13210] stasis.c: Topic 'cache:71/channel:1629282830.61': 0x7f0c10037770 destroyed [Aug 18 10:34:20] DEBUG[14718] http.c: match request [ari/channels/213101] with handler [httpstatus] len 10 [Aug 18 10:34:20] DEBUG[14716] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: pushing 0x7f0c200663a0(Recorder/ARI-00000039;2) [Aug 18 10:34:20] DEBUG[13318] stasis.c: Topic 'cache:104/channel:1629282832.89': 0x7f0c2808d730 destroyed [Aug 18 10:34:20] DEBUG[13318] stasis.c: Destroying topic. name: channel:1629282832.89, detail: [Aug 18 10:34:20] DEBUG[13318] stasis.c: Topic 'channel:1629282832.89': 0x7f0c2808cdb0 destroyed [Aug 18 10:34:20] DEBUG[14718] http.c: match request [ari/channels/213101] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14720] http.c: match request [ari/channels/213099] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14720] http.c: match request [ari/channels/213099] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'cache:331/channel:213099': 0x7f0c840881b0 destroyed [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #129)) [Aug 18 10:34:20] DEBUG[14717] http.c: match request [ari/channels/213094] with handler [phoneprov] len 9 [Aug 18 10:34:20] DEBUG[13210] stasis.c: Destroying topic. name: channel:1629282830.61, detail: [Aug 18 10:34:20] DEBUG[14706] stasis.c: Creating topic. name: channel:213159, detail: [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: channel:213099, detail: [Aug 18 10:34:20] DEBUG[14708] rtp_engine.c: RTP instance '0x7f0c7012f9f0' is setup and ready to go [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) ICE stopped [Aug 18 10:34:20] DEBUG[14708] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:20] DEBUG[14708] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:20] DEBUG[14708] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:20] DEBUG[14708] res_rtp_asterisk.c: (0x7f0c7012f9f0) RTCP setup on RTP instance [Aug 18 10:34:20] VERBOSE[14708] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:20] DEBUG[14708] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:20] DEBUG[14708] chan_sip.c: SIP call-id changed from '0c34a1e010e388a561dcf0f45bc608f5@127.0.1.1:5060' to '3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060' [Aug 18 10:34:20] DEBUG[14708] stasis.c: Creating topic. name: channel:213162, detail: [Aug 18 10:34:20] DEBUG[14708] stasis.c: Topic 'channel:213162': 0x7f0c7003e1f0 created [Aug 18 10:34:20] DEBUG[14720] http.c: Match made with [ari] [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116897@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9 Max-Forwards: 70 From: ;tag=as5acf84f3 To: Contact: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168069357 1168069357 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18198 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14717] http.c: match request [ari/channels/213094] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14717] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'channel:213099': 0x7f0c84139df0 destroyed [Aug 18 10:34:20] DEBUG[13210] stasis.c: Topic 'channel:1629282830.61': 0x7f0c10030630 destroyed [Aug 18 10:34:20] DEBUG[14718] http.c: match request [ari/channels/213101] with handler [ari] len 3 [Aug 18 10:34:20] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:11', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000087', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213094', '')] [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282860.454, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.454': 0x7f0c30070380 created [Aug 18 10:34:20] DEBUG[20545] stasis.c: Creating topic. name: cache:522/channel:1629282860.454, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:522/channel:1629282860.454': 0x7f0c30012dc0 created [Aug 18 10:34:20] DEBUG[14708] stasis.c: Creating topic. name: cache:521/channel:213162, detail: [Aug 18 10:34:20] DEBUG[14708] stasis.c: Topic 'cache:521/channel:213162': 0x7f0c70007420 created [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Finding handler for 213102 [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Finding handler for channels/213099 [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking channels create: Didn't match 213102 [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14706] stasis.c: Topic 'channel:213159': 0x7f0c400c6fe0 created [Aug 18 10:34:20] DEBUG[14706] stasis.c: Creating topic. name: cache:523/channel:213159, detail: [Aug 18 10:34:20] DEBUG[14706] stasis.c: Topic 'cache:523/channel:213159': 0x7f0c400ca430 created [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #7 (2) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #7)) [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116901@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b Max-Forwards: 70 From: ;tag=as3a1d6e7b To: Contact: Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1147590624 1147590624 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6611ms with no response [Aug 18 10:34:20] WARNING[20585] chan_sip.c: Hanging up call 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (5) INVITE - 5 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #49)) [Aug 18 10:34:20] DEBUG[14719] res_ari.c: Checking channels externalMedia: Didn't match 213102 [Aug 18 10:34:20] DEBUG[14715] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pushing 0x7f0c24057ae0(Announcer/ARI-00000038;2) [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1372568962 1372568962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11634 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:20] DEBUG[14718] http.c: Match made with [ari] [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:20] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14408] channel.c: Channel 0x7f0c7405a610 'SIP/zvonobot-00000095' hanging up. Refs: 2 [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Finding handler for channels/213101 [Aug 18 10:34:20] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:20] DEBUG[14719] res_ari.c: No explicit handler found for 213102. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: bridge '87d87304-31e6-4326-b367-680423189269': is 1 interested in calls_0 [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[20620] stasis/app.c: bridge '87d87304-31e6-4326-b367-680423189269' unsubscribed from calls_0 [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] WARNING[14444] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000031;1 [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: cache:41/bridge:87d87304-31e6-4326-b367-680423189269, detail: [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'cache:41/bridge:87d87304-31e6-4326-b367-680423189269': 0x7f0cb4012030 destroyed [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14310] bridge_channel.c: Bridge is returning 0x7f0c80044670(Recorder/ARI-0000002e;2) to write format slin [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Finding handler for 213101 [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking channels create: Didn't match 213101 [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] DEBUG[14718] res_ari.c: Checking channels externalMedia: Didn't match 213101 [Aug 18 10:34:20] DEBUG[14718] res_ari.c: No explicit handler found for 213101. Using wildcard channelId. [Aug 18 10:34:20] DEBUG[20620] stasis.c: Destroying topic. name: bridge:87d87304-31e6-4326-b367-680423189269, detail: [Aug 18 10:34:20] DEBUG[14310] channel.c: Channel Recorder/ARI-0000002e;2 setting write format path: slin -> slin [Aug 18 10:34:20] DEBUG[14310] channel.c: Channel 0x7f0c80042760 'Recorder/ARI-0000002e;2' hanging up. Refs: 2 [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: cache:522/channel:1629282860.454, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'cache:522/channel:1629282860.454': 0x7f0c30012dc0 destroyed [Aug 18 10:34:20] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282860.454, detail: [Aug 18 10:34:20] DEBUG[20545] stasis.c: Topic 'channel:1629282860.454': 0x7f0c30070380 destroyed [Aug 18 10:34:20] DEBUG[20620] stasis.c: Topic 'bridge:87d87304-31e6-4326-b367-680423189269': 0x7f0cb4027c50 destroyed [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Finding handler for 213099 [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking channels create: Didn't match 213099 [Aug 18 10:34:20] DEBUG[14711] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:20] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:20] DEBUG[14717] res_ari.c: Finding handler for channels/213094 [Aug 18 10:34:20] DEBUG[14716] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:20] DEBUG[14717] res_ari.c: Finding handler for channels [Aug 18 10:34:20] DEBUG[14717] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:20] DEBUG[14717] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:20] DEBUG[14717] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:20] WARNING[14717] logger: Log queue threshold (1000) exceeded. Discarding new messages. [Aug 18 10:34:20] VERBOSE[14716] bridge_channel.c: Channel Recorder/ARI-00000039;2 joined 'simple_bridge' stasis-bridge <61075423-3ee2-4d60-8382-ee99e654a5be> [Aug 18 10:34:20] DEBUG[14720] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:20] WARNING[20531] logger: Logging resumed. 782 messages discarded. [Aug 18 10:34:20] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b9cf791;received=159.65.48.104 From: ;tag=as57703f31 To: ;tag=as5bdb9e2d Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1779078100 1779078100 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13598 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b9cf791;received=159.65.48.104 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57703f31 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5bdb9e2d [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:20] DEBUG[13419] res_rtp_asterisk.c: (0x7f0c2003ba40) RTP 0x7f0c20044340 -- Received packet from 178.62.121.41:16216, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[13419] res_rtp_asterisk.c: (0x7f0c2003ba40) RTP 0x7f0c20044340 -- Received packet from 178.62.121.41:16216, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[13419] res_rtp_asterisk.c: (0x7f0c2003ba40) RTP 0x7f0c20044340 -- Received packet from 178.62.121.41:16216, dropping due to strict RTP protection. [Aug 18 10:34:20] DEBUG[14735] http.c: HTTP opening session. Top level [Aug 18 10:34:20] DEBUG[14604] res_stasis.c: calls_0: Subscribing to 213153 [Aug 18 10:34:20] DEBUG[14734] channel.c: Channel Announcer/ARI-00000038;1 setting write format path: gsm -> slin [Aug 18 10:34:20] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:20] DEBUG[14735] http.c: HTTP Request URI is /ari/bridges/cb82e822-34ce-4cab-8b96-97e1b95e246e/addChannel?channel=1629282854.367%2Crobot_212965 [Aug 18 10:34:20] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1779078100 1779078100 IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13598 RTP/AVP 0 8 101 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 (Checking To) --From tag as57703f31 --To-tag as5bdb9e2d [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Got SDP version 1779078100 and unique parts [root 1779078100 IN IP4 178.62.121.41] [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1779078100 1779078100 IN IP4 178.62.121.41... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:21] DEBUG[14604] stasis/app.c: Channel '213153' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Outgoing Call for 79821116887 [Aug 18 10:34:21] DEBUG[14735] http.c: match request [ari/bridges/cb82e822-34ce-4cab-8b96-97e1b95e246e/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:21] DEBUG[14604] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14735] http.c: match request [ari/bridges/cb82e822-34ce-4cab-8b96-97e1b95e246e/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:21] DEBUG[14604] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14735] http.c: match request [ari/bridges/cb82e822-34ce-4cab-8b96-97e1b95e246e/addChannel] with handler [ari] len 3 [Aug 18 10:34:21] VERBOSE[14732] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 [Aug 18 10:34:21] DEBUG[14735] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] DEBUG[13416] res_rtp_asterisk.c: (0x7f0c30074490) RTP 0x7f0c30075fe0 -- Received packet from 178.62.121.41:19448, dropping due to strict RTP protection. [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Finding handler for bridges/cb82e822-34ce-4cab-8b96-97e1b95e246e/addChannel [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Finding handler for cb82e822-34ce-4cab-8b96-97e1b95e246e [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:21] DEBUG[14735] res_ari.c: No explicit handler found for cb82e822-34ce-4cab-8b96-97e1b95e246e. Using wildcard bridgeId. [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Finding handler for addChannel [Aug 18 10:34:21] DEBUG[14735] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:21] DEBUG[14735] stasis/control.c: 1629282854.367: Sending channel add_to_bridge command [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:21] DEBUG[14734] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:21] VERBOSE[14734] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[14519] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP got report of 76 bytes from 178.62.121.41:12381 [Aug 18 10:34:21] VERBOSE[14519] res_rtp_asterisk.c: 0x7f0c980a2f60 -- Strict RTP learning complete - Locking on source address 178.62.121.41:12380 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:21] DEBUG[14732] stasis/app.c: Channel 'robot_213022' is 2 interested in calls_0 [Aug 18 10:34:21] DEBUG[14570] bridge_roles.c: Roles did not exist on channel Snoop/212965-00000012 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] DEBUG[14570] stasis/control.c: 1629282854.367: Adding to bridge cb82e822-34ce-4cab-8b96-97e1b95e246e [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:21] DEBUG[14570] stasis/app.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14737] bridge_channel.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: 0x7f0c80044670(Snoop/212965-00000012) is joining [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:21] VERBOSE[14736] chan_sip.c: Audio is at 10240 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00b650) ICE set role failed; no ice instance [Aug 18 10:34:21] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:21] DEBUG[14737] bridge_channel.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: pushing 0x7f0c80044670(Snoop/212965-00000012) [Aug 18 10:34:21] VERBOSE[14737] bridge_channel.c: Channel Snoop/212965-00000012 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00b650) RTCP setting address on RTP instance [Aug 18 10:34:21] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c1c03e660 -- Strict RTP learning after remote address set to: 178.62.121.41:13598 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:13598 [Aug 18 10:34:21] VERBOSE[14736] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0130e08) from 0x7f0c147e2330 to 0x7f0c1c00b828 [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282861.467, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.467': 0x7f0c30162520 created [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: cache:537/channel:1629282861.467, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:537/channel:1629282861.467': 0x7f0c30162f80 created [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: cache:537/channel:1629282861.467, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:537/channel:1629282861.467': 0x7f0c30162f80 destroyed [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282861.467, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.467': 0x7f0c30162520 destroyed [Aug 18 10:34:21] DEBUG[14737] bridge_native_rtp.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' can not use native RTP bridge as two channels are required [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0130ae8) from 0x7f0c147e2330 to 0x7f0c1c00b828 [Aug 18 10:34:21] DEBUG[14737] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] DEBUG[14737] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14737] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14737] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14737] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e is already using the new technology. [Aug 18 10:34:21] DEBUG[14737] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: 0x7f0c80044670(Snoop/212965-00000012) is joining simple_bridge technology [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00985a8) from 0x7f0c147e2330 to 0x7f0c1c00b828 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00b650) RTCP ignoring duplicate property [Aug 18 10:34:21] VERBOSE[14736] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:21] VERBOSE[14736] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000000', '', 'Stasis', 'calls_0', 32, 27, 'ANSWERED', 3, '', '212964', '')] [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:21] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000032 setting read format path: alaw -> alaw [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000032 setting write format path: alaw -> alaw [Aug 18 10:34:21] DEBUG[14735] stasis/control.c: robot_212965: Sending channel add_to_bridge command [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00b650) DTLS - ast_rtp_activate rtp=0x7f0c1c03e660 - setup and perform DTLS' [Aug 18 10:34:21] DEBUG[14570] stasis/app.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' is 2 interested in calls_0 [Aug 18 10:34:21] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c03e660) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Initializing initreq for method INVITE - callid 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116887@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c03e660) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Strict routing enforced for session 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117026@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61b9c719 Max-Forwards: 70 From: ;tag=as57703f31 To: ;tag=as5bdb9e2d Contact: Call-ID: 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (4) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116904@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face Max-Forwards: 70 From: ;tag=as1a5706e7 To: Contact: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 641212001 641212001 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13764 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 3 [ 52]: From: ;tag=as7b595413 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282861.468, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.468': 0x7f0c30162600 created [Aug 18 10:34:21] VERBOSE[13255] dial.c: SIP/zvonobot-00000032 answered [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: cache:538/channel:1629282861.468, detail: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Session timer started: 12 - 321ca4232d3de4fa6bb441530619a899@159.65.48.104:5060 1768000ms [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #130 (5) BYE - 8 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:21] VERBOSE[13255] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000032 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #130)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: BYE sip:79821117076@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963 Max-Forwards: 70 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117076@178.62.121.41:5060", nonce="11c410aa", response="b5fe7439b49314fb4f0f18ebd5c2f549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 6 [ 60]: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[13255] stasis/app.c: Channel '213014' is 2 interested in calls_0 [Aug 18 10:34:21] VERBOSE[13255] res_rtp_asterisk.c: 0x7f0c1c03e660 -- Strict RTP switching to RTP target address 178.62.121.41:13598 as source [Aug 18 10:34:21] DEBUG[13255] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:21] DEBUG[13255] channel.c: Channel SIP/zvonobot-00000032 setting read format path: ulaw -> alaw [Aug 18 10:34:21] DEBUG[13255] channel.c: Channel SIP/zvonobot-00000032 setting write format path: alaw -> ulaw [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:538/channel:1629282861.468': 0x7f0c30006720 created [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14596] channel.c: Channel 0x7f0cb4024d90 'SIP/zvonobot-000000b5' allocated [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[14596] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: cache:538/channel:1629282861.468, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:538/channel:1629282861.468': 0x7f0c30006720 destroyed [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282861.468, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.468': 0x7f0c30162600 destroyed [Aug 18 10:34:21] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:47', '"" <>', '', 's', 'default', 'Snoop/212964-00000000', 'UnicastRTP/127.0.0.1:50394-0x7f0c18009d50', 'Stasis', 'calls_0', 22, 22, 'ANSWERED', 3, '', '1629282827.33', '')] [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] VERBOSE[14736] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116887@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 Max-Forwards: 70 From: ;tag=as7b595413 To: Contact: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 184815596 184815596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10240 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Aug 18 10:34:21] DEBUG[14736] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a05a098;received=159.65.48.104 From: ;tag=as733c3052 To: ;tag=as5cb7ffc5 Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 709745446 709745446 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12818 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:21] DEBUG[14731] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 [Aug 18 10:34:21] DEBUG[14731] stasis/control.c: robot_212965: Adding to bridge cb82e822-34ce-4cab-8b96-97e1b95e246e [Aug 18 10:34:21] DEBUG[14731] stasis/app.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' is 3 interested in calls_0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:21] DEBUG[14596] res_stasis.c: calls_0: Subscribing to 213150 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0a05a098;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[14738] bridge_channel.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: 0x7f0c2404d900(UnicastRTP/127.0.0.1:50065-0x7f0c900475d0) is joining [Aug 18 10:34:21] DEBUG[14596] stasis/app.c: Channel '213150' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14596] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14596] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as733c3052 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5cb7ffc5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 709745446 709745446 IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12818 RTP/AVP 0 8 101 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 (Checking To) --From tag as733c3052 --To-tag as5cb7ffc5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:21] DEBUG[14738] bridge_channel.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: pushing 0x7f0c2404d900(UnicastRTP/127.0.0.1:50065-0x7f0c900475d0) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Got SDP version 709745446 and unique parts [root 709745446 IN IP4 178.62.121.41] [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 709745446 709745446 IN IP4 178.62.121.41... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:21] VERBOSE[14736] dial.c: Called zvonobot/79821116887 [Aug 18 10:34:21] VERBOSE[14738] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Outgoing Call for 79821116890 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:21] DEBUG[14738] bridge_native_rtp.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e'. Checking compatability for channels 'Snoop/212965-00000012' and 'UnicastRTP/127.0.0.1:50065-0x7f0c900475d0' [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14738] bridge_native_rtp.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' can not use native RTP bridge as could not get details [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] DEBUG[14738] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] VERBOSE[14739] chan_sip.c: Audio is at 16040 [Aug 18 10:34:21] DEBUG[14738] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] VERBOSE[14739] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] DEBUG[14738] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14738] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:21] DEBUG[14738] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e is already using the new technology. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:21] DEBUG[14738] bridge.c: Bridge cb82e822-34ce-4cab-8b96-97e1b95e246e: 0x7f0c2404d900(UnicastRTP/127.0.0.1:50065-0x7f0c900475d0) is joining simple_bridge technology [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:21] VERBOSE[14739] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:21] VERBOSE[14739] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Initializing initreq for method INVITE - callid 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116890@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 setting read format path: slin16 -> slin16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel Snoop/212965-00000012 setting write format path: slin16 -> slin [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel Snoop/212965-00000012 setting read format path: slin -> slin16 [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 setting write format path: slin16 -> slin16 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90008240) ICE set role failed; no ice instance [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 3 [ 52]: From: ;tag=as148e9e8b [Aug 18 10:34:21] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90008240) RTCP setting address on RTP instance [Aug 18 10:34:21] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c9000c310 -- Strict RTP learning after remote address set to: 178.62.121.41:12818 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12818 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00e4268) from 0x7f0c147e2330 to 0x7f0c90008418 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 6 [ 60]: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb004f688) from 0x7f0c147e2330 to 0x7f0c90008418 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00900a8) from 0x7f0c147e2330 to 0x7f0c90008418 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90008240) RTCP ignoring duplicate property [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:21] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000d setting read format path: alaw -> alaw [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[14735] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:21] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000d setting write format path: alaw -> alaw [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90008240) DTLS - ast_rtp_activate rtp=0x7f0c9000c310 - setup and perform DTLS' [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9000c310) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9000c310) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Strict routing enforced for session 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117061@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b1f4a3a Max-Forwards: 70 From: ;tag=as733c3052 To: ;tag=as5cb7ffc5 Contact: Call-ID: 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14735] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14731] stasis/app.c: Bridge 'cb82e822-34ce-4cab-8b96-97e1b95e246e' is 4 interested in calls_0 [Aug 18 10:34:21] VERBOSE[12948] dial.c: SIP/zvonobot-0000000d answered [Aug 18 10:34:21] DEBUG[14584] channel.c: Channel 0x7f0ca811b900 'SIP/zvonobot-000000b6' allocated [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[14584] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] DEBUG[14740] http.c: HTTP opening session. Top level [Aug 18 10:34:21] VERBOSE[12948] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000000d [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (4) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:21] DEBUG[14740] http.c: HTTP Request URI is /ari/bridges/48ea2adf-d916-4f02-8d4c-5c853a6c83c0/addChannel?channel=1629282854.369%2Crobot_213022 [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:21] DEBUG[14740] http.c: match request [ari/bridges/48ea2adf-d916-4f02-8d4c-5c853a6c83c0/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP got report of 100 bytes from 178.62.121.41:10695 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:21] DEBUG[14740] http.c: match request [ari/bridges/48ea2adf-d916-4f02-8d4c-5c853a6c83c0/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 849904962 849904962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14740] http.c: match request [ari/bridges/48ea2adf-d916-4f02-8d4c-5c853a6c83c0/addChannel] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14740] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[12948] stasis/app.c: Channel '212979' is 2 interested in calls_0 [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Finding handler for bridges/48ea2adf-d916-4f02-8d4c-5c853a6c83c0/addChannel [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Session timer started: 92 - 33e3eb2f592f706b0a085a7f06be7869@159.65.48.104:5060 1768000ms [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] VERBOSE[12948] res_rtp_asterisk.c: 0x7f0c9000c310 -- Strict RTP switching to RTP target address 178.62.121.41:12818 as source [Aug 18 10:34:21] DEBUG[12948] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:21] DEBUG[14738] channel.c: Channel UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 setting write format path: slin -> slin16 [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50065-0x7f0c900475d0 - start 1629282860.829306 answer 1629282860.918635 end 1629282861.199316 dur 0.370 bill 0.280 dispo ANSWERED [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:21] DEBUG[12948] channel.c: Channel SIP/zvonobot-0000000d setting read format path: ulaw -> alaw [Aug 18 10:34:21] DEBUG[12948] channel.c: Channel SIP/zvonobot-0000000d setting write format path: alaw -> ulaw [Aug 18 10:34:21] DEBUG[14738] res_rtp_asterisk.c: (0x7f0c900475d0) RTP ooh, format changed from none to slin16 [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] VERBOSE[14739] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Finding handler for 48ea2adf-d916-4f02-8d4c-5c853a6c83c0 [Aug 18 10:34:21] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #80 [Aug 18 10:34:21] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:21] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:21] DEBUG[14584] res_stasis.c: calls_0: Subscribing to 213144 [Aug 18 10:34:21] DEBUG[14584] stasis/app.c: Channel '213144' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14740] res_ari.c: No explicit handler found for 48ea2adf-d916-4f02-8d4c-5c853a6c83c0. Using wildcard bridgeId. [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Finding handler for addChannel [Aug 18 10:34:21] DEBUG[14739] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Outgoing Call for 79821116896 [Aug 18 10:34:21] DEBUG[14584] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14584] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14740] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:21] DEBUG[14740] stasis/control.c: 1629282854.369: Sending channel add_to_bridge command [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] DEBUG[14569] bridge_roles.c: Roles did not exist on channel Snoop/213022-00000013 [Aug 18 10:34:21] DEBUG[14569] stasis/control.c: 1629282854.369: Adding to bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0 [Aug 18 10:34:21] DEBUG[14569] stasis/app.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14742] bridge_channel.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: 0x7f0c840862b0(Snoop/213022-00000013) is joining [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] VERBOSE[14741] chan_sip.c: Audio is at 12660 [Aug 18 10:34:21] VERBOSE[14741] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] VERBOSE[14741] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] VERBOSE[14741] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] VERBOSE[14739] dial.c: Called zvonobot/79821116890 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64;received=159.65.48.104 From: ;tag=as293a990c To: ;tag=as401cd277 Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48bad81c" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Initializing initreq for method INVITE - callid 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as293a990c [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116896@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as401cd277 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14742] bridge_channel.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: pushing 0x7f0c840862b0(Snoop/213022-00000013) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 3 [ 52]: From: ;tag=as67ede665 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48bad81c" [Aug 18 10:34:21] VERBOSE[14742] bridge_channel.c: Channel Snoop/213022-00000013 joined 'simple_bridge' stasis-bridge <48ea2adf-d916-4f02-8d4c-5c853a6c83c0> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 6 [ 60]: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 (Checking To) --From tag as293a990c --To-tag as401cd277 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[14742] bridge_native_rtp.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' can not use native RTP bridge as two channels are required [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (5) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:21] DEBUG[14742] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1764277553 1764277553 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15278 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14742] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14742] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14742] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14742] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0 is already using the new technology. [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[14742] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: 0x7f0c840862b0(Snoop/213022-00000013) is joining simple_bridge technology [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[14741] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #113 [Aug 18 10:34:21] DEBUG[14741] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (4) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116898@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 Max-Forwards: 70 From: ;tag=as35c0c7eb To: Contact: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 144689560 144689560 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116887@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 Max-Forwards: 70 From: ;tag=as7b595413 To: Contact: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 184815596 184815596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10240 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14740] stasis/control.c: robot_213022: Sending channel add_to_bridge command [Aug 18 10:34:21] DEBUG[14569] stasis/app.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' is 2 interested in calls_0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[14529] res_rtp_asterisk.c: 0x2c17ba0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:15860 [Aug 18 10:34:21] VERBOSE[14741] dial.c: Called zvonobot/79821116896 [Aug 18 10:34:21] DEBUG[14591] channel.c: Channel 0x7f0cb0169950 'SIP/zvonobot-000000b7' allocated [Aug 18 10:34:21] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[14591] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] DEBUG[14591] res_stasis.c: calls_0: Subscribing to 213149 [Aug 18 10:34:21] DEBUG[14591] stasis/app.c: Channel '213149' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Outgoing Call for 79821116891 [Aug 18 10:34:21] DEBUG[14591] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14591] http.c: HTTP closing session. Top level [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1070485752 1070485752 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15860 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42732cd2;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4ca0b4bd [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5f99fd9e [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[14732] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14732] stasis/control.c: robot_213022: Adding to bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0 [Aug 18 10:34:21] DEBUG[14732] stasis/app.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' is 3 interested in calls_0 [Aug 18 10:34:21] DEBUG[14529] res_rtp_asterisk.c: (0x2c14110) RTCP got report of 76 bytes from 178.62.121.41:15861 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[14745] bridge_channel.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: 0x7f0c20073c60(UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0) is joining [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] VERBOSE[14744] chan_sip.c: Audio is at 15752 [Aug 18 10:34:21] VERBOSE[14744] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] VERBOSE[14744] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] VERBOSE[14744] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[13354] app.c: One waitfor failed, trying another [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14587] channel.c: Channel 0x7f0c9808a1b0 'SIP/zvonobot-000000b9' allocated [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Initializing initreq for method INVITE - callid 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1070485752 1070485752 IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116891@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15860 RTP/AVP 0 8 101 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 3 [ 52]: From: ;tag=as4d536c24 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 6 [ 60]: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] VERBOSE[14744] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #121 [Aug 18 10:34:21] DEBUG[14744] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 (Checking To) --From tag as4ca0b4bd --To-tag as5f99fd9e [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14745] bridge_channel.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: pushing 0x7f0c20073c60(UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0) [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Strict routing enforced for session 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:21] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:21] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117037@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d5e8c67 Max-Forwards: 70 From: ;tag=as4ca0b4bd To: ;tag=as5f99fd9e Contact: Call-ID: 65566df87e9df09f6e52e7d87b74dc8e@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #129 (4) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #129)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116897@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9 Max-Forwards: 70 From: ;tag=as5acf84f3 To: Contact: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168069357 1168069357 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18198 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #7 (4) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #7)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116901@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b Max-Forwards: 70 From: ;tag=as3a1d6e7b To: Contact: Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1147590624 1147590624 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (2) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116887@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 Max-Forwards: 70 From: ;tag=as7b595413 To: Contact: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 184815596 184815596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10240 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] VERBOSE[14744] dial.c: Called zvonobot/79821116891 [Aug 18 10:34:21] DEBUG[14587] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] VERBOSE[14745] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 joined 'simple_bridge' stasis-bridge <48ea2adf-d916-4f02-8d4c-5c853a6c83c0> [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 - start 1629282860.793504 answer 1629282860.925769 end 1629282861.492626 dur 0.699 bill 0.566 dispo ANSWERED [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289;received=159.65.48.104 From: ;tag=as09899d91 To: ;tag=as2041a243 Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="732ec2da" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[14745] bridge_native_rtp.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0'. Checking compatability for channels 'Snoop/213022-00000013' and 'UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0' [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04e52289;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as09899d91 [Aug 18 10:34:21] DEBUG[14745] bridge_native_rtp.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' can not use native RTP bridge as could not get details [Aug 18 10:34:21] DEBUG[14745] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2041a243 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="732ec2da" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[14745] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14745] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14745] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14745] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0 is already using the new technology. [Aug 18 10:34:21] DEBUG[14745] bridge.c: Bridge 48ea2adf-d916-4f02-8d4c-5c853a6c83c0: 0x7f0c20073c60(UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0) is joining simple_bridge technology [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 setting read format path: slin16 -> slin16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 (Checking To) --From tag as09899d91 --To-tag as2041a243 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2d4029193f64fb721f43803f29facceb@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel Snoop/213022-00000013 setting write format path: slin16 -> slin [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel Snoop/213022-00000013 setting read format path: slin -> slin16 [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 setting write format path: slin16 -> slin16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (5) INVITE - 5 [Aug 18 10:34:21] DEBUG[14603] channel.c: Channel 0x7f0c08046230 'SIP/zvonobot-000000b8' allocated [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116907@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772 Max-Forwards: 70 From: ;tag=as6c7cfd27 To: Contact: Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 680344710 680344710 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14603] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:21] DEBUG[14607] channel.c: Channel 0x7f0c200cad60 'SIP/zvonobot-000000ba' allocated [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[14607] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14732] stasis/app.c: Bridge '48ea2adf-d916-4f02-8d4c-5c853a6c83c0' is 4 interested in calls_0 [Aug 18 10:34:21] DEBUG[14740] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:21] DEBUG[14740] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14747] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14747] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:21] DEBUG[14747] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[14749] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14749] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:21] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:21] DEBUG[14747] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[14749] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[14749] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[14747] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14603] res_stasis.c: calls_0: Subscribing to 213152 [Aug 18 10:34:21] DEBUG[14603] stasis/app.c: Channel '213152' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14603] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14603] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14747] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 5 instead [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[14749] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:21] DEBUG[14749] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[14587] res_stasis.c: calls_0: Subscribing to 213145 [Aug 18 10:34:21] DEBUG[14587] stasis/app.c: Channel '213145' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Outgoing Call for 79821116888 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb;received=159.65.48.104 From: ;tag=as4a9f4c08 To: ;tag=as3424bdf4 Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="794b29c0" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4a9f4c08 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3424bdf4 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="794b29c0" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 (Checking To) --From tag as4a9f4c08 --To-tag as3424bdf4 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116932@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58322cbb Max-Forwards: 70 From: ;tag=as4a9f4c08 To: Contact: Call-ID: 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #109 (6) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #109)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116924@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7781c5db Max-Forwards: 70 From: ;tag=as1e86ce5f To: Contact: Call-ID: 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 602516760 602516760 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12662 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0;received=159.65.48.104 From: ;tag=as080d6dff To: ;tag=as5181b3f0 Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78be2a15" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41f352a0;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as080d6dff [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5181b3f0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Outgoing Call for 79821116895 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[13556] res_rtp_asterisk.c: (0x7f0c7c020d90) RTCP got report of 100 bytes from 178.62.121.41:14675 [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14607] res_stasis.c: calls_0: Subscribing to 213146 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[14745] channel.c: Channel UnicastRTP/127.0.0.1:50199-0x7f0c880b67b0 setting write format path: slin -> slin16 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] DEBUG[14607] stasis/app.c: Channel '213146' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[14587] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14587] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78be2a15" [Aug 18 10:34:21] DEBUG[14607] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14607] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] DEBUG[14745] res_rtp_asterisk.c: (0x7f0c880b67b0) RTP ooh, format changed from none to slin16 [Aug 18 10:34:21] DEBUG[14462] channel.c: Channel 0x7f0c38071710 'UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0' allocated [Aug 18 10:34:21] DEBUG[14462] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:21] VERBOSE[14462] res_rtp_asterisk.c: 0x7f0c380940e0 -- Strict RTP learning after remote address set to: 127.0.0.1:50211 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] VERBOSE[14752] chan_sip.c: Audio is at 10316 [Aug 18 10:34:21] VERBOSE[14752] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] VERBOSE[14752] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] VERBOSE[14752] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Initializing initreq for method INVITE - callid 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116888@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 3 [ 52]: From: ;tag=as1c2a52a2 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 6 [ 60]: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] VERBOSE[14752] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1216361947 1216361947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #85 [Aug 18 10:34:21] DEBUG[14752] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] VERBOSE[14752] dial.c: Called zvonobot/79821116888 [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] DEBUG[14747] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] DEBUG[14462] res_stasis.c: calls_0: Subscribing to robot_212991 [Aug 18 10:34:21] DEBUG[14462] stasis/app.c: Channel 'robot_212991' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[14462] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[14747] stasis.c: Creating topic. name: bridge:b9e5c782-ffd8-4b13-9c95-cdea191c152d, detail: [Aug 18 10:34:21] DEBUG[14462] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 (Checking To) --From tag as080d6dff --To-tag as5181b3f0 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 083441fd621bd040753e952c5d9a1860@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Outgoing Call for 79821116894 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:21] DEBUG[14747] stasis.c: Topic 'bridge:b9e5c782-ffd8-4b13-9c95-cdea191c152d': 0x7f0c800137d0 created [Aug 18 10:34:21] DEBUG[14747] stasis.c: Creating topic. name: cache:539/bridge:b9e5c782-ffd8-4b13-9c95-cdea191c152d, detail: [Aug 18 10:34:21] DEBUG[14747] stasis.c: Topic 'cache:539/bridge:b9e5c782-ffd8-4b13-9c95-cdea191c152d': 0x7f0c800271f0 created [Aug 18 10:34:21] VERBOSE[14753] chan_sip.c: Audio is at 16832 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] VERBOSE[14753] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] DEBUG[14747] bridge_native_rtp.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d' can not use native RTP bridge as two channels are required [Aug 18 10:34:21] DEBUG[14747] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] DEBUG[14747] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14747] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:21] DEBUG[14747] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14747] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: calling simple_bridge technology constructor [Aug 18 10:34:21] DEBUG[14747] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: calling simple_bridge technology start [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (2) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:21] DEBUG[14747] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] VERBOSE[14753] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] VERBOSE[14754] chan_sip.c: Audio is at 17760 [Aug 18 10:34:21] DEBUG[14747] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] VERBOSE[14756] dial.c: Called 127.0.0.1:50211 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:21] VERBOSE[14753] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[14757] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[14757] http.c: HTTP Request URI is /ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/addChannel?channel=213014 [Aug 18 10:34:21] VERBOSE[14754] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:21] DEBUG[14757] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[14757] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[14757] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/addChannel] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14757] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[14749] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] VERBOSE[14754] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:21] DEBUG[14749] stasis.c: Creating topic. name: bridge:1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be, detail: [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Finding handler for bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/addChannel [Aug 18 10:34:21] DEBUG[14749] stasis.c: Topic 'bridge:1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be': 0x7f0c8c106960 created [Aug 18 10:34:21] DEBUG[14749] stasis.c: Creating topic. name: cache:540/bridge:1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be, detail: [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb;received=159.65.48.104 From: ;tag=as3bdaa5eb To: ;tag=as2ebcbfcc Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b57b9fd" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Initializing initreq for method INVITE - callid 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116895@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 3 [ 52]: From: ;tag=as123f2352 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 6 [ 60]: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:21 GMT [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] VERBOSE[14753] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2052644047 2052644047 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16832 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Aug 18 10:34:21] DEBUG[14749] stasis.c: Topic 'cache:540/bridge:1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be': 0x7f0c8c038a30 created [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14753] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3bdaa5eb [Aug 18 10:34:21] VERBOSE[14754] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] VERBOSE[14753] dial.c: Called zvonobot/79821116895 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2ebcbfcc [Aug 18 10:34:21] DEBUG[14657] app.c: One waitfor failed, trying another [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Finding handler for b9e5c782-ffd8-4b13-9c95-cdea191c152d [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:21] DEBUG[14757] res_ari.c: No explicit handler found for b9e5c782-ffd8-4b13-9c95-cdea191c152d. Using wildcard bridgeId. [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Finding handler for addChannel [Aug 18 10:34:21] DEBUG[14757] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[14757] stasis/control.c: 213014: Sending channel add_to_bridge command [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14749] bridge_native_rtp.c: Bridge '1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be' can not use native RTP bridge as two channels are required [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14749] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b57b9fd" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] DEBUG[14749] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[14749] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:21] DEBUG[13255] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000032 [Aug 18 10:34:21] DEBUG[13255] stasis/control.c: 213014: Adding to bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d [Aug 18 10:34:21] DEBUG[13255] stasis/app.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d' is 1 interested in calls_0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 (Checking To) --From tag as3bdaa5eb --To-tag as2ebcbfcc [Aug 18 10:34:21] DEBUG[14759] bridge_channel.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: 0x7f0c180860e0(SIP/zvonobot-00000032) is joining [Aug 18 10:34:21] DEBUG[14749] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14749] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: calling simple_bridge technology constructor [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:21] VERBOSE[14756] dial.c: UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0 answered [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14749] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: calling simple_bridge technology start [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116931@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK418393bb Max-Forwards: 70 From: ;tag=as3bdaa5eb To: Contact: Call-ID: 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[14759] bridge_channel.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: pushing 0x7f0c180860e0(SIP/zvonobot-00000032) [Aug 18 10:34:21] VERBOSE[14759] bridge_channel.c: Channel SIP/zvonobot-00000032 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Initializing initreq for method INVITE - callid 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14759] bridge_native_rtp.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d' can not use native RTP bridge as two channels are required [Aug 18 10:34:21] DEBUG[14759] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:21] DEBUG[14759] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116894@178.62.121.41 SIP/2.0 [Aug 18 10:34:21] DEBUG[14759] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051 [Aug 18 10:34:21] DEBUG[14759] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:21] DEBUG[14759] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d is already using the new technology. [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6432ms with no response [Aug 18 10:34:21] DEBUG[14759] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: 0x7f0c180860e0(SIP/zvonobot-00000032) is joining simple_bridge technology [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Hanging up call 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:21] DEBUG[14749] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14749] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:21] DEBUG[14760] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14760] http.c: HTTP Request URI is /ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/addChannel?channel=212979 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (2) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:21] DEBUG[14760] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[13255] stasis/app.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d' is 2 interested in calls_0 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6513ms with no response [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Hanging up call 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14757] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 3 [ 52]: From: ;tag=as3e829f44 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:21] VERBOSE[14756] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0 [Aug 18 10:34:21] DEBUG[14761] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14759] res_rtp_asterisk.c: (0x7f0c1c00b650) RTP changing ssrc from 995532785 to 1287917332 due to a source change [Aug 18 10:34:21] DEBUG[14466] channel.c: Channel 0x7f0c280d8530 'SIP/zvonobot-00000097' hanging up. Refs: 2 [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000098 - start 1629282855.215475 answer 0.000000 end 1629282861.817646 dur 6.602 bill 1629282861.817 dispo NO ANSWER [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000097 - start 1629282855.068621 answer 0.000000 end 1629282861.824313 dur 6.755 bill 1629282861.824 dispo NO ANSWER [Aug 18 10:34:21] DEBUG[14757] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1216361947 1216361947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (6) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116922@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3 Max-Forwards: 70 From: ;tag=as7674a2b1 To: Contact: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 489493290 489493290 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12284 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (2) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14468] channel.c: Channel 0x7f0c340f0030 'SIP/zvonobot-00000098' hanging up. Refs: 2 [Aug 18 10:34:21] DEBUG[14761] http.c: HTTP Request URI is /ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/record?name=213014_FacccbxlHWyHfCuOUZlihAHrKMCUFPsZ&format=wav [Aug 18 10:34:21] DEBUG[14761] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/record] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[14760] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[14760] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/addChannel] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14760] http.c: Match made with [ari] [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1;received=159.65.48.104 From: ;tag=as3a399b13 To: ;tag=as3bcd3559 Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68776d7e" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a399b13 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3bcd3559 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14761] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/record] with handler [phoneprov] len 9 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Finding handler for bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/addChannel [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Finding handler for bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14761] http.c: match request [ari/bridges/b9e5c782-ffd8-4b13-9c95-cdea191c152d/record] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68776d7e" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 (Checking To) --From tag as3a399b13 --To-tag as3bcd3559 [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 6 [ 60]: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[14761] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '63e4041b488585c57e57de141ed1835f@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116935@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1 Max-Forwards: 70 From: ;tag=as3a399b13 To: Contact: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Finding handler for 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6410ms with no response [Aug 18 10:34:21] DEBUG[14760] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Hanging up call 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:21] DEBUG[14760] res_ari.c: No explicit handler found for 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be. Using wildcard bridgeId. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:21] WARNING[14760] logger: Log queue threshold (1000) exceeded. Discarding new messages. [Aug 18 10:34:21] DEBUG[14754] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:21] WARNING[20531] logger: Logging resumed. 100 messages discarded. [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43;received=159.65.48.104 From: ;tag=as126e0733 To: ;tag=as7887a81a Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b521c7e" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as126e0733 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7887a81a [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b521c7e" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 (Checking To) --From tag as126e0733 --To-tag as7887a81a [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '137837c51322c444587a45b5059337ee@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116934@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43 Max-Forwards: 70 From: ;tag=as126e0733 To: Contact: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Finding handler for 7182caa2-2514-4ffa-b2c8-5bbdd18d9794 [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:21] DEBUG[14763] res_ari.c: No explicit handler found for 7182caa2-2514-4ffa-b2c8-5bbdd18d9794. Using wildcard bridgeId. [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Finding handler for addChannel [Aug 18 10:34:21] DEBUG[14763] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:21] DEBUG[14763] ari/resource_bridges.c: Found non-stasis '1629282851.322' [Aug 18 10:34:21] DEBUG[14763] http.c: HTTP keeping session open. status_code:422 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as2a29680d Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1600602803 1600602803 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12714 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2a29680d [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[14762] bridge_channel.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: pushing 0x7f0ca80df8f0(SIP/zvonobot-0000000d) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:21] DEBUG[14599] channel.c: Channel 0x2c9e500 'SIP/zvonobot-000000bb' allocated [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:21] DEBUG[14763] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1600602803 1600602803 IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12714 RTP/AVP 0 8 101 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as2a29680d [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6397ms with no response [Aug 18 10:34:21] WARNING[20585] chan_sip.c: Hanging up call 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14599] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #104 (6) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #104)) [Aug 18 10:34:21] DEBUG[14475] channel.c: Channel 0x7f0c3c1318b0 'SIP/zvonobot-00000099' hanging up. Refs: 2 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482061060 1482061060 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[14051] chan_sip.c: Hangup call SIP/zvonobot-00000072, SIP callid 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14051] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:21] DEBUG[14051] res_rtp_asterisk.c: (0x7f0ca8034280) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[14051] res_rtp_asterisk.c: (0x7f0ca8034280) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[14051] channel.c: Channel 0x7f0ca80e0110 'SIP/zvonobot-00000072' destroying [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[14615] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000099 - start 1629282855.444669 answer 0.000000 end 1629282861.927388 dur 6.482 bill 1629282861.927 dispo NO ANSWER [Aug 18 10:34:21] DEBUG[14615] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:21] DEBUG[20620] stasis/app.c: channel '213075': is 0 interested in calls_0 [Aug 18 10:34:21] DEBUG[20620] stasis/app.c: channel '213075' unsubscribed from calls_0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2;received=159.65.48.104 From: ;tag=as452417ef To: ;tag=as57995f03 Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as452417ef [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as57995f03 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 (Checking To) --From tag as452417ef --To-tag as57995f03 [Aug 18 10:34:21] DEBUG[14765] http.c: HTTP opening session. Top level [Aug 18 10:34:21] DEBUG[14051] stasis.c: Destroying topic. name: cache:284/channel:213075, detail: [Aug 18 10:34:21] DEBUG[14051] stasis.c: Topic 'cache:284/channel:213075': 0x7f0ca807a110 destroyed [Aug 18 10:34:21] DEBUG[14051] stasis.c: Destroying topic. name: channel:213075, detail: [Aug 18 10:34:21] DEBUG[14051] stasis.c: Topic 'channel:213075': 0x7f0ca806b390 destroyed [Aug 18 10:34:21] DEBUG[14765] http.c: HTTP Request URI is /ari/channels/213075 [Aug 18 10:34:21] DEBUG[14765] http.c: match request [ari/channels/213075] with handler [httpstatus] len 10 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282861.470, detail: [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Stopping retransmission on '4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.470': 0x7f0c300b9b30 created [Aug 18 10:34:21] DEBUG[20545] stasis.c: Creating topic. name: cache:542/channel:1629282861.470, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:542/channel:1629282861.470': 0x7f0c301019c0 created [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: cache:542/channel:1629282861.470, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'cache:542/channel:1629282861.470': 0x7f0c301019c0 destroyed [Aug 18 10:34:21] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282861.470, detail: [Aug 18 10:34:21] DEBUG[20545] stasis.c: Topic 'channel:1629282861.470': 0x7f0c300b9b30 destroyed [Aug 18 10:34:21] VERBOSE[14762] bridge_channel.c: Channel SIP/zvonobot-0000000d joined 'simple_bridge' stasis-bridge <1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be> [Aug 18 10:34:21] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000072', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213075', '')] [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[14765] http.c: match request [ari/channels/213075] with handler [phoneprov] len 9 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117014@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK155478c2 Max-Forwards: 70 From: ;tag=as452417ef To: ;tag=as57995f03 Contact: Call-ID: 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:21] DEBUG[14765] http.c: match request [ari/channels/213075] with handler [ari] len 3 [Aug 18 10:34:21] DEBUG[14765] http.c: Match made with [ari] [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Destroying SIP dialog 6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Finding handler for channels/213075 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '6e44566f0d7fab971f2020243e30ec00@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:21] VERBOSE[13423] dial.c: SIP/zvonobot-00000043 is busy [Aug 18 10:34:21] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000043 - start 1629282834.789048 answer 0.000000 end 1629282861.961055 dur 27.172 bill 1629282861.961 dispo BUSY [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8034280) DTLS stop [Aug 18 10:34:21] DEBUG[13423] channel.c: Channel 0x7f0c280bd370 'SIP/zvonobot-00000043' hanging up. Refs: 2 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Finding handler for channels [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8034280) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8034280) DTLS srtp - stopped timeout timer' [Aug 18 10:34:21] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8034280) ICE RTP transport deallocating [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:21] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:21] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca8034280' [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (2) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1216361947 1216361947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (1) INVITE - 5 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116894@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051 Max-Forwards: 70 From: ;tag=as3e829f44 To: Contact: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1459261102 1459261102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456;received=159.65.48.104 From: ;tag=as307f6396 To: ;tag=as3a0db143 Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37ef5652" Content-Length: 0 <-------------> [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456;received=159.65.48.104 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as307f6396 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3a0db143 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37ef5652" [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:21] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 (Checking To) --From tag as307f6396 --To-tag as3a0db143 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Finding handler for 213075 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking channels create: Didn't match 213075 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:21] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: Checking channels externalMedia: Didn't match 213075 [Aug 18 10:34:21] DEBUG[14765] res_ari.c: No explicit handler found for 213075. Using wildcard channelId. [Aug 18 10:34:21] DEBUG[14474] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:21] DEBUG[14474] http.c: HTTP closing session. Top level [Aug 18 10:34:21] DEBUG[14605] channel.c: Channel 0x7f0c18105090 'SIP/zvonobot-000000bc' allocated [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:21] DEBUG[14605] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:22] DEBUG[14605] res_stasis.c: calls_0: Subscribing to 213148 [Aug 18 10:34:22] DEBUG[14605] stasis/app.c: Channel '213148' is 1 interested in calls_0 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Outgoing Call for 79821116892 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:22] DEBUG[14605] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:22] DEBUG[14605] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[14762] bridge_native_rtp.c: Bridge '1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be' can not use native RTP bridge as two channels are required [Aug 18 10:34:22] DEBUG[14762] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:22] DEBUG[14762] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 From: ;tag=as08e169d8 To: ;tag=as58d484f0 Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1044364027 1044364027 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12716 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5df64882;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[14762] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08e169d8 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as58d484f0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] VERBOSE[14766] chan_sip.c: Audio is at 12378 [Aug 18 10:34:22] VERBOSE[14766] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:22] VERBOSE[14766] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[14762] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:22] VERBOSE[14766] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:22] DEBUG[14762] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be is already using the new technology. [Aug 18 10:34:22] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1044364027 1044364027 IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[14762] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: 0x7f0ca80df8f0(SIP/zvonobot-0000000d) is joining simple_bridge technology [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Initializing initreq for method INVITE - callid 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116892@178.62.121.41 SIP/2.0 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12716 RTP/AVP 0 8 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 (Checking To) --From tag as08e169d8 --To-tag as58d484f0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Strict routing enforced for session 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 3 [ 52]: From: ;tag=as0a953bb4 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 6 [ 60]: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:22 GMT [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] VERBOSE[14766] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116892@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 Max-Forwards: 70 From: ;tag=as0a953bb4 To: Contact: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 772240936 772240936 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #123 [Aug 18 10:34:22] DEBUG[14766] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117074@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67d4902f Max-Forwards: 70 From: ;tag=as08e169d8 To: ;tag=as58d484f0 Contact: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6420ms with no response [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Hanging up call 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[14766] dial.c: Called zvonobot/79821116892 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (3) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14477] channel.c: Channel 0x7f0c84094380 'SIP/zvonobot-0000009b' hanging up. Refs: 2 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 From: ;tag=as5e4952fc To: ;tag=as228e10c3 Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1109970970 1109970970 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3d22f647;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5e4952fc [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as228e10c3 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1109970970 1109970970 IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14956 RTP/AVP 0 8 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 (Checking To) --From tag as5e4952fc --To-tag as228e10c3 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Strict routing enforced for session 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117019@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1a425357 Max-Forwards: 70 From: ;tag=as5e4952fc To: ;tag=as228e10c3 Contact: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14769] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14769] http.c: HTTP Request URI is /ari/channels/213164?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116876&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14769] http.c: match request [ari/channels/213164] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14769] http.c: match request [ari/channels/213164] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14769] http.c: match request [ari/channels/213164] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14769] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14769] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14762] res_rtp_asterisk.c: (0x7f0c90008240) RTP changing ssrc from 169665551 to 61687684 due to a source change [Aug 18 10:34:22] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000009b - start 1629282855.546772 answer 0.000000 end 1629282862.039862 dur 6.493 bill 1629282862.039 dispo NO ANSWER [Aug 18 10:34:22] DEBUG[14760] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:22] DEBUG[14760] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[12948] stasis/app.c: Bridge '1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be' is 2 interested in calls_0 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Finding handler for channels/213164 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14770] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14770] http.c: HTTP Request URI is /ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/record?name=212979_JmGFaAKGpLdhTmasRWipwagxgDaiCpdz&format=wav [Aug 18 10:34:22] DEBUG[14770] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/record] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14770] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/record] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14770] http.c: match request [ari/bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/record] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14770] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as2a29680d Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1600602803 1600602803 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12714 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Finding handler for bridges/1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be/record [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14773] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Finding handler for bridges [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Finding handler for 213164 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking channels create: Didn't match 213164 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14769] res_ari.c: Checking channels externalMedia: Didn't match 213164 [Aug 18 10:34:22] DEBUG[14769] res_ari.c: No explicit handler found for 213164. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:22] DEBUG[14773] http.c: HTTP Request URI is /ari/channels/213167?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116873&callerId=74950493843 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2a29680d [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1600602803 1600602803 IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12714 RTP/AVP 0 8 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as2a29680d [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (2) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2052644047 2052644047 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16832 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '398559732fb8625271bea90231b90490@159.65.48.104:5060' [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (6) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 40468863 40468863 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15158 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Destroying SIP dialog 398559732fb8625271bea90231b90490@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '398559732fb8625271bea90231b90490@159.65.48.104:5060' Method: BYE [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS stop [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74010590) ICE RTP transport deallocating [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c74010590' [Aug 18 10:34:22] DEBUG[14773] http.c: match request [ari/channels/213167] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14773] http.c: match request [ari/channels/213167] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14773] http.c: match request [ari/channels/213167] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Finding handler for 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14770] res_ari.c: No explicit handler found for 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be. Using wildcard bridgeId. [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Finding handler for record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:22] DEBUG[14770] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:22] DEBUG[14775] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14616] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:22] DEBUG[14616] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[14599] res_stasis.c: calls_0: Subscribing to 213151 [Aug 18 10:34:22] DEBUG[14599] stasis/app.c: Channel '213151' is 1 interested in calls_0 [Aug 18 10:34:22] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14599] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:22] DEBUG[14773] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14775] http.c: HTTP Request URI is /ari/channels/213165?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116875&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14773] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14770] stasis.c: Creating topic. name: channel:1629282862.471, detail: [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14775] http.c: match request [ari/channels/213165] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14775] http.c: match request [ari/channels/213165] with handler [phoneprov] len 9 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2;received=159.65.48.104 From: ;tag=as6400b9b5 To: ;tag=as56577292 Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ee1fa31" Content-Length: 0 <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6400b9b5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56577292 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ee1fa31" [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 (Checking To) --From tag as6400b9b5 --To-tag as56577292 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116923@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2 Max-Forwards: 70 From: ;tag=as6400b9b5 To: Contact: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (3) INVITE - 5 [Aug 18 10:34:22] DEBUG[14782] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14599] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14770] stasis.c: Topic 'channel:1629282862.471': 0x7f0c08075de0 created [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Finding handler for channels/213167 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14770] stasis.c: Creating topic. name: cache:543/channel:1629282862.471, detail: [Aug 18 10:34:22] DEBUG[14770] stasis.c: Topic 'cache:543/channel:1629282862.471': 0x7f0c08084370 created [Aug 18 10:34:22] DEBUG[14775] http.c: match request [ari/channels/213165] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[13556] audiohook.c: Audiohook 0x7f0ca0038440 has stale audio in its factories. Flushing them both [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14775] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14782] http.c: HTTP Request URI is /ari/channels/213168?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116872&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Finding handler for 213167 [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking channels create: Didn't match 213167 [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14773] res_ari.c: Checking channels externalMedia: Didn't match 213167 [Aug 18 10:34:22] DEBUG[14773] res_ari.c: No explicit handler found for 213167. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Outgoing Call for 79821116889 [Aug 18 10:34:22] DEBUG[14775] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14782] http.c: match request [ari/channels/213168] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14782] http.c: match request [ari/channels/213168] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14782] http.c: match request [ari/channels/213168] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc;received=159.65.48.104 From: ;tag=as2a62315e To: ;tag=as3efd5c26 Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42031c6d" Content-Length: 0 <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2a62315e [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3efd5c26 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42031c6d" [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 (Checking To) --From tag as2a62315e --To-tag as3efd5c26 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116918@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc Max-Forwards: 70 From: ;tag=as2a62315e To: Contact: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (1) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116892@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 Max-Forwards: 70 From: ;tag=as0a953bb4 To: Contact: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 772240936 772240936 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (2) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116894@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051 Max-Forwards: 70 From: ;tag=as3e829f44 To: Contact: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1459261102 1459261102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6390ms with no response [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Hanging up call 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14782] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14784] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14782] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Finding handler for channels/213168 [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Finding handler for 213168 [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking channels create: Didn't match 213168 [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14782] res_ari.c: Checking channels externalMedia: Didn't match 213168 [Aug 18 10:34:22] DEBUG[14782] res_ari.c: No explicit handler found for 213168. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14784] http.c: HTTP Request URI is /ari/channels/213166?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116874&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Finding handler for channels/213165 [Aug 18 10:34:22] DEBUG[14479] channel.c: Channel 0x7f0c78022bf0 'SIP/zvonobot-0000009c' hanging up. Refs: 2 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ffed886;received=159.65.48.104 From: ;tag=as41697fdf To: ;tag=as53a098fa Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1334571769 1334571769 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15546 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6ffed886;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as41697fdf [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as53a098fa [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[14624] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:22] DEBUG[14624] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[14784] http.c: match request [ari/channels/213166] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000009c - start 1629282855.589527 answer 0.000000 end 1629282862.174204 dur 6.584 bill 1629282862.174 dispo NO ANSWER [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:22] DEBUG[14784] http.c: match request [ari/channels/213166] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14785] http.c: HTTP opening session. Top level [Aug 18 10:34:22] VERBOSE[14779] chan_sip.c: Audio is at 16966 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14784] http.c: match request [ari/channels/213166] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14784] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14784] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Finding handler for channels/213166 [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Finding handler for 213166 [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking channels create: Didn't match 213166 [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14784] res_ari.c: Checking channels externalMedia: Didn't match 213166 [Aug 18 10:34:22] DEBUG[14784] res_ari.c: No explicit handler found for 213166. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14785] http.c: HTTP Request URI is /ari/channels/213170?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116870&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14785] http.c: match request [ari/channels/213170] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14785] http.c: match request [ari/channels/213170] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[14664] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] DEBUG[14786] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14786] http.c: HTTP Request URI is /ari/channels/213169?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116871&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14785] http.c: match request [ari/channels/213170] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14785] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1334571769 1334571769 IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15546 RTP/AVP 0 8 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 (Checking To) --From tag as41697fdf --To-tag as53a098fa [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Finding handler for 213165 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking channels create: Didn't match 213165 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14775] res_ari.c: Checking channels externalMedia: Didn't match 213165 [Aug 18 10:34:22] DEBUG[14775] res_ari.c: No explicit handler found for 213165. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] VERBOSE[14779] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:22] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[14785] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14786] http.c: match request [ari/channels/213169] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: Allocating new SIP dialog for 63a7e8c16d8826a16f4ef1f516d65ab9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14782] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2001b180' [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) RTP allocated port 12396 [Aug 18 10:34:22] DEBUG[14786] http.c: match request [ari/channels/213169] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14786] http.c: match request [ari/channels/213169] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: Allocating new SIP dialog for 1984d7154c9a2aaf21a388ca436db396@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14769] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c10088ef0' [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) RTP allocated port 17132 [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE creating session 0.0.0.0:17132 (17132) [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '135d1bf70165f2237445fa857e02663b@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Got SDP version 1334571769 and unique parts [root 1334571769 IN IP4 178.62.121.41] [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1334571769 1334571769 IN IP4 178.62.121.41... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000a6f0) ICE set role failed; no ice instance [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE creating session 0.0.0.0:12396 (12396) [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE create [Aug 18 10:34:22] DEBUG[14786] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14786] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Finding handler for channels/213169 [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Finding handler for 213169 [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking channels create: Didn't match 213169 [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14786] res_ari.c: Checking channels externalMedia: Didn't match 213169 [Aug 18 10:34:22] DEBUG[14786] res_ari.c: No explicit handler found for 213169. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE create [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE add system candidates [Aug 18 10:34:22] DEBUG[14769] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14769] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE add candidate: 159.65.48.104:17132, 2130706431 [Aug 18 10:34:22] DEBUG[14769] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14769] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE add candidate: 10.131.0.10:17132, 2130706431 [Aug 18 10:34:22] DEBUG[14769] rtp_engine.c: RTP instance '0x7f0c10088ef0' is setup and ready to go [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE stopped [Aug 18 10:34:22] DEBUG[14769] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:22] DEBUG[14769] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14769] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:22] DEBUG[14769] res_rtp_asterisk.c: (0x7f0c10088ef0) RTCP setup on RTP instance [Aug 18 10:34:22] VERBOSE[14769] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:22] DEBUG[14769] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14769] chan_sip.c: SIP call-id changed from '1984d7154c9a2aaf21a388ca436db396@127.0.1.1:5060' to '2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060' [Aug 18 10:34:22] DEBUG[14769] stasis.c: Creating topic. name: channel:213164, detail: [Aug 18 10:34:22] DEBUG[14769] stasis.c: Topic 'channel:213164': 0x7f0c10072d60 created [Aug 18 10:34:22] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: Allocating new SIP dialog for 014f6e5b449afc5c4a60cc386cfed8cb@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14784] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c098760' [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) RTP allocated port 19192 [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE creating session 0.0.0.0:19192 (19192) [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE create [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE add system candidates [Aug 18 10:34:22] DEBUG[14784] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14784] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE add candidate: 159.65.48.104:19192, 2130706431 [Aug 18 10:34:22] DEBUG[14784] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14784] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE add candidate: 10.131.0.10:19192, 2130706431 [Aug 18 10:34:22] DEBUG[14784] rtp_engine.c: RTP instance '0x7f0c2c098760' is setup and ready to go [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) ICE stopped [Aug 18 10:34:22] DEBUG[14784] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:22] DEBUG[14784] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14784] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:22] DEBUG[14784] res_rtp_asterisk.c: (0x7f0c2c098760) RTCP setup on RTP instance [Aug 18 10:34:22] VERBOSE[14784] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:22] DEBUG[14784] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14784] chan_sip.c: SIP call-id changed from '014f6e5b449afc5c4a60cc386cfed8cb@127.0.1.1:5060' to '7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060' [Aug 18 10:34:22] DEBUG[14784] stasis.c: Creating topic. name: channel:213166, detail: [Aug 18 10:34:22] DEBUG[14784] stasis.c: Topic 'channel:213166': 0x7f0c2c0e3530 created [Aug 18 10:34:22] DEBUG[14784] stasis.c: Creating topic. name: cache:545/channel:213166, detail: [Aug 18 10:34:22] DEBUG[14784] stasis.c: Topic 'cache:545/channel:213166': 0x7f0c2c0e3fb0 created [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Finding handler for channels/213170 [Aug 18 10:34:22] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE add system candidates [Aug 18 10:34:22] DEBUG[14782] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14782] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE add candidate: 159.65.48.104:12396, 2130706431 [Aug 18 10:34:22] DEBUG[14782] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14782] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE add candidate: 10.131.0.10:12396, 2130706431 [Aug 18 10:34:22] DEBUG[14782] rtp_engine.c: RTP instance '0x7f0c2001b180' is setup and ready to go [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) ICE stopped [Aug 18 10:34:22] DEBUG[14782] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:22] DEBUG[14782] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14782] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:22] DEBUG[14782] res_rtp_asterisk.c: (0x7f0c2001b180) RTCP setup on RTP instance [Aug 18 10:34:22] VERBOSE[14782] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:22] DEBUG[14782] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14782] chan_sip.c: SIP call-id changed from '63a7e8c16d8826a16f4ef1f516d65ab9@127.0.1.1:5060' to '1e454c24705858e9259d323c756ca026@159.65.48.104:5060' [Aug 18 10:34:22] DEBUG[14782] stasis.c: Creating topic. name: channel:213168, detail: [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 848, ms is 73 [Aug 18 10:34:22] DEBUG[14769] stasis.c: Creating topic. name: cache:544/channel:213164, detail: [Aug 18 10:34:22] DEBUG[14769] stasis.c: Topic 'cache:544/channel:213164': 0x7f0c100737e0 created [Aug 18 10:34:22] DEBUG[14787] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14787] http.c: HTTP Request URI is /ari/channels/213171?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116869&callerId=74950493843 [Aug 18 10:34:22] DEBUG[14787] http.c: match request [ari/channels/213171] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14787] http.c: match request [ari/channels/213171] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14787] http.c: match request [ari/channels/213171] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14787] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14787] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Finding handler for channels/213171 [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Finding handler for 213171 [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking channels create: Didn't match 213171 [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14787] res_ari.c: Checking channels externalMedia: Didn't match 213171 [Aug 18 10:34:22] DEBUG[14787] res_ari.c: No explicit handler found for 213171. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14727] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] VERBOSE[14779] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:22] DEBUG[14789] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14789] http.c: HTTP Request URI is /ari/channels/213173?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116867&callerId=74950493843 [Aug 18 10:34:22] DEBUG[13483] res_rtp_asterisk.c: (0x7f0c3c052f40) RTP audio difference is 816, ms is 71 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTCP setting address on RTP instance [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Finding handler for 213170 [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking channels create: Didn't match 213170 [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14785] res_ari.c: Checking channels externalMedia: Didn't match 213170 [Aug 18 10:34:22] DEBUG[14785] res_ari.c: No explicit handler found for 213170. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14788] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 848, ms is 73 [Aug 18 10:34:22] DEBUG[14782] stasis.c: Topic 'channel:213168': 0x7f0c200e3120 created [Aug 18 10:34:22] DEBUG[13441] audiohook.c: Audiohook 0x7f0c280c2020 has stale audio in its factories. Flushing them both [Aug 18 10:34:22] DEBUG[14789] http.c: match request [ari/channels/213173] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14789] http.c: match request [ari/channels/213173] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14789] http.c: match request [ari/channels/213173] with handler [ari] len 3 [Aug 18 10:34:22] VERBOSE[14779] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:22] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0ca000e480 -- Strict RTP learning after remote address set to: 178.62.121.41:15546 [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[14789] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14789] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14788] http.c: HTTP Request URI is /ari/channels/213172?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116868&callerId=74950493843 [Aug 18 10:34:22] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 9 instead [Aug 18 10:34:22] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:15546 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00e44b8) from 0x7f0c147e2330 to 0x7f0ca000a8c8 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0046ed8) from 0x7f0c147e2330 to 0x7f0ca000a8c8 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00f0908) from 0x7f0c147e2330 to 0x7f0ca000a8c8 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTCP ignoring duplicate property [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:22] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000f setting read format path: alaw -> alaw [Aug 18 10:34:22] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000000f setting write format path: alaw -> alaw [Aug 18 10:34:22] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Initializing initreq for method INVITE - callid 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116889@178.62.121.41 SIP/2.0 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 3 [ 52]: From: ;tag=as1ac06673 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 6 [ 60]: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:22 GMT [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] VERBOSE[14779] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116889@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 Max-Forwards: 70 From: ;tag=as1ac06673 To: Contact: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 722698992 722698992 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #79 [Aug 18 10:34:22] DEBUG[14779] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14788] http.c: match request [ari/channels/213172] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14782] stasis.c: Creating topic. name: cache:546/channel:213168, detail: [Aug 18 10:34:22] DEBUG[14782] stasis.c: Topic 'cache:546/channel:213168': 0x7f0c200bd740 created [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Finding handler for channels/213173 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000a6f0) DTLS - ast_rtp_activate rtp=0x7f0ca000e480 - setup and perform DTLS' [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] VERBOSE[14779] dial.c: Called zvonobot/79821116889 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000e480) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca000e480) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:22] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Strict routing enforced for session 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117060@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK201f9ca3 Max-Forwards: 70 From: ;tag=as41697fdf To: ;tag=as53a098fa Contact: Call-ID: 135d1bf70165f2237445fa857e02663b@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14788] http.c: match request [ari/channels/213172] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: Allocating new SIP dialog for 539afc29044d66e8395ae20e1784743a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14786] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c340a2dd0' [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) RTP allocated port 12786 [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE creating session 0.0.0.0:12786 (12786) [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE create [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE add system candidates [Aug 18 10:34:22] DEBUG[14786] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14786] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE add candidate: 159.65.48.104:12786, 2130706431 [Aug 18 10:34:22] DEBUG[14786] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14786] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE add candidate: 10.131.0.10:12786, 2130706431 [Aug 18 10:34:22] DEBUG[14786] rtp_engine.c: RTP instance '0x7f0c340a2dd0' is setup and ready to go [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) ICE stopped [Aug 18 10:34:22] DEBUG[14786] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:22] DEBUG[14786] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14786] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:22] DEBUG[14786] res_rtp_asterisk.c: (0x7f0c340a2dd0) RTCP setup on RTP instance [Aug 18 10:34:22] VERBOSE[14786] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:22] DEBUG[14786] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Finding handler for channels [Aug 18 10:34:22] VERBOSE[12959] dial.c: SIP/zvonobot-0000000f answered [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:22] VERBOSE[12959] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000000f [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (3) INVITE - 5 [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14788] http.c: match request [ari/channels/213172] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:22] DEBUG[12959] stasis/app.c: Channel '212980' is 2 interested in calls_0 [Aug 18 10:34:22] VERBOSE[12959] res_rtp_asterisk.c: 0x7f0ca000e480 -- Strict RTP switching to RTP target address 178.62.121.41:15546 as source [Aug 18 10:34:22] DEBUG[12959] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:22] DEBUG[12959] channel.c: Channel SIP/zvonobot-0000000f setting read format path: ulaw -> alaw [Aug 18 10:34:22] DEBUG[12959] channel.c: Channel SIP/zvonobot-0000000f setting write format path: alaw -> ulaw [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:22] DEBUG[14775] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Finding handler for 213173 [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking channels create: Didn't match 213173 [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14789] res_ari.c: Checking channels externalMedia: Didn't match 213173 [Aug 18 10:34:22] DEBUG[14789] res_ari.c: No explicit handler found for 213173. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:22] DEBUG[14773] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14788] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14775] chan_sip.c: Allocating new SIP dialog for 0a65612148b8870675a360126e2a316a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14773] chan_sip.c: Allocating new SIP dialog for 012043cb1f261c761b2b7df21543de52@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14773] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c139d70' [Aug 18 10:34:22] DEBUG[14773] res_rtp_asterisk.c: (0x7f0c1c139d70) RTP allocated port 13196 [Aug 18 10:34:22] DEBUG[14773] res_rtp_asterisk.c: (0x7f0c1c139d70) ICE creating session 0.0.0.0:13196 (13196) [Aug 18 10:34:22] DEBUG[14773] res_rtp_asterisk.c: (0x7f0c1c139d70) ICE create [Aug 18 10:34:22] DEBUG[14773] res_rtp_asterisk.c: (0x7f0c1c139d70) ICE add system candidates [Aug 18 10:34:22] DEBUG[14773] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14773] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14786] chan_sip.c: SIP call-id changed from '539afc29044d66e8395ae20e1784743a@127.0.1.1:5060' to '7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060' [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #132 (5) INVITE - 5 [Aug 18 10:34:22] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14788] http.c: HTTP consuming request body [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Finding handler for channels/213172 [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Finding handler for channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Finding handler for 213172 [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking channels create: Didn't match 213172 [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14788] res_ari.c: Checking channels externalMedia: Didn't match 213172 [Aug 18 10:34:22] DEBUG[14788] res_ari.c: No explicit handler found for 213172. Using wildcard channelId. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #132)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116906@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74 Max-Forwards: 70 From: ;tag=as5e0e197a To: Contact: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 306689819 306689819 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10106 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6497ms with no response [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Hanging up call 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14775] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c180435e0' [Aug 18 10:34:22] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:22] DEBUG[14785] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14785] chan_sip.c: Allocating new SIP dialog for 67651b1a30892e783ad6e48850ef3320@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14785] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280cdcb0' [Aug 18 10:34:22] DEBUG[14785] res_rtp_asterisk.c: (0x7f0c280cdcb0) RTP allocated port 14652 [Aug 18 10:34:22] DEBUG[14785] res_rtp_asterisk.c: (0x7f0c280cdcb0) ICE creating session 0.0.0.0:14652 (14652) [Aug 18 10:34:22] DEBUG[14785] res_rtp_asterisk.c: (0x7f0c280cdcb0) ICE create [Aug 18 10:34:22] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14484] channel.c: Channel 0x7f0c8008bb10 'SIP/zvonobot-0000009f' hanging up. Refs: 2 [Aug 18 10:34:22] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000009f - start 1629282855.900700 answer 0.000000 end 1629282862.461367 dur 6.560 bill 1629282862.461 dispo NO ANSWER [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[14786] stasis.c: Creating topic. name: channel:213169, detail: [Aug 18 10:34:22] DEBUG[14786] stasis.c: Topic 'channel:213169': 0x7f0c340daa60 created [Aug 18 10:34:22] DEBUG[14786] stasis.c: Creating topic. name: cache:547/channel:213169, detail: [Aug 18 10:34:22] DEBUG[14786] stasis.c: Topic 'cache:547/channel:213169': 0x7f0c340db4c0 created [Aug 18 10:34:22] DEBUG[14785] res_rtp_asterisk.c: (0x7f0c280cdcb0) ICE add system candidates [Aug 18 10:34:22] DEBUG[14785] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (5) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116903@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09 Max-Forwards: 70 From: ;tag=as41f91965 To: Contact: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1172096761 1172096761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:22] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6507ms with no response [Aug 18 10:34:22] WARNING[20585] chan_sip.c: Hanging up call 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14773] res_rtp_asterisk.c: (0x7f0c1c139d70) ICE add candidate: 159.65.48.104:13196, 2130706431 [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) RTP allocated port 10394 [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE creating session 0.0.0.0:10394 (10394) [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE create [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE add system candidates [Aug 18 10:34:22] DEBUG[14775] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14775] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE add candidate: 159.65.48.104:10394, 2130706431 [Aug 18 10:34:22] DEBUG[14775] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14775] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE add candidate: 10.131.0.10:10394, 2130706431 [Aug 18 10:34:22] DEBUG[14775] rtp_engine.c: RTP instance '0x7f0c180435e0' is setup and ready to go [Aug 18 10:34:22] DEBUG[14775] res_rtp_asterisk.c: (0x7f0c180435e0) ICE stopped [Aug 18 10:34:22] DEBUG[14775] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:22] DEBUG[14775] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14791] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14791] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:22] DEBUG[14620] channel.c: Channel 0x7f0c74079320 'Announcer/ARI-00000042;1' allocated [Aug 18 10:34:22] DEBUG[14483] channel.c: Channel 0x7f0c940ed7b0 'SIP/zvonobot-0000009d' hanging up. Refs: 2 [Aug 18 10:34:22] DEBUG[14482] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:22] WARNING[20620] logger: Log queue threshold (1000) exceeded. Discarding new messages. [Aug 18 10:34:22] DEBUG[14775] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:22] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000009d - start 1629282855.803831 answer 0.000000 end 1629282862.478933 dur 6.675 bill 1629282862.478 dispo NO ANSWER [Aug 18 10:34:22] DEBUG[14785] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] DEBUG[14791] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:22] WARNING[20531] logger: Logging resumed. 712 messages discarded. [Aug 18 10:34:22] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:22] DEBUG[20545] stasis.c: Destroying topic. name: cache:560/channel:1629282862.487, detail: [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'cache:560/channel:1629282862.487': 0x7f0c300a65d0 destroyed [Aug 18 10:34:22] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282862.487, detail: [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'channel:1629282862.487': 0x7f0c300bdf10 destroyed [Aug 18 10:34:22] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:02', '"" <>', '', 's', 'default', 'Snoop/213007-0000000d', 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00', 'Stasis', 'calls_0', 16, 16, 'ANSWERED', 3, '', '1629282842.212', '')] [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 From: ;tag=as18b114f0 To: ;tag=as342fd06d Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 392402988 392402988 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14534 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as18b114f0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as342fd06d [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:22] DEBUG[14526] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:22] DEBUG[14526] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 392402988 392402988 IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14534 RTP/AVP 0 8 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: = Looking for Call ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 (Checking To) --From tag as18b114f0 --To-tag as342fd06d [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Stopping retransmission on '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Got SDP version 392402988 and unique parts [root 392402988 IN IP4 178.62.121.41] [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 392402988 392402988 IN IP4 178.62.121.41... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) ICE set role failed; no ice instance [Aug 18 10:34:22] DEBUG[14799] bridge_native_rtp.c: Bridge 'c8381fea-1239-48c9-a6e3-1d9ad7226cf1' can not use native RTP bridge as two channels are required [Aug 18 10:34:22] DEBUG[14799] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:22] DEBUG[14799] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:22] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:22] DEBUG[14799] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Outgoing Call for 79821116893 [Aug 18 10:34:22] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282862.489, detail: [Aug 18 10:34:22] DEBUG[14788] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:22] DEBUG[14788] chan_sip.c: Allocating new SIP dialog for 66bea07450b6898d59e5b9a461a6d094@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:22] DEBUG[14788] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c3c159c70' [Aug 18 10:34:22] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) RTP allocated port 14666 [Aug 18 10:34:22] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE creating session 0.0.0.0:14666 (14666) [Aug 18 10:34:22] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE create [Aug 18 10:34:22] DEBUG[14799] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'channel:1629282862.489': 0x7f0c300bdf10 created [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) RTCP setting address on RTP instance [Aug 18 10:34:22] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c2002a6f0 -- Strict RTP learning after remote address set to: 178.62.121.41:14534 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14534 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb006ec68) from 0x7f0c147e2330 to 0x7f0c20028d78 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00d0cd8) from 0x7f0c147e2330 to 0x7f0c20028d78 [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0098558) from 0x7f0c147e2330 to 0x7f0c20028d78 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) RTCP ignoring duplicate property [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:22] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000034 setting read format path: alaw -> alaw [Aug 18 10:34:22] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000034 setting write format path: alaw -> alaw [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) DTLS - ast_rtp_activate rtp=0x7f0c2002a6f0 - setup and perform DTLS' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2002a6f0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2002a6f0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:22] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Strict routing enforced for session 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:22] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:22] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117024@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK056af0b4 Max-Forwards: 70 From: ;tag=as18b114f0 To: ;tag=as342fd06d Contact: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE add system candidates [Aug 18 10:34:22] DEBUG[20545] stasis.c: Creating topic. name: cache:562/channel:1629282862.489, detail: [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (4) INVITE - 5 [Aug 18 10:34:22] VERBOSE[13271] dial.c: SIP/zvonobot-00000034 answered [Aug 18 10:34:22] DEBUG[14799] bridge.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1 is already using the new technology. [Aug 18 10:34:22] DEBUG[14799] bridge.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1: 0x7f0ca4110b70(SIP/zvonobot-0000000f) is joining simple_bridge technology [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:22] DEBUG[14788] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:22] DEBUG[14788] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:22] VERBOSE[13271] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000034 [Aug 18 10:34:22] DEBUG[13271] stasis/app.c: Channel '213016' is 2 interested in calls_0 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:22] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'cache:562/channel:1629282862.489': 0x7f0c300e4c40 created [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (5) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 849904962 849904962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (5) INVITE - 5 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116898@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 Max-Forwards: 70 From: ;tag=as35c0c7eb To: Contact: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 144689560 144689560 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE add candidate: 159.65.48.104:14666, 2130706431 [Aug 18 10:34:22] DEBUG[20545] stasis.c: Destroying topic. name: cache:562/channel:1629282862.489, detail: [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'cache:562/channel:1629282862.489': 0x7f0c300e4c40 destroyed [Aug 18 10:34:22] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282862.489, detail: [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'channel:1629282862.489': 0x7f0c300bdf10 destroyed [Aug 18 10:34:22] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:48', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001f', '', 'Stasis', 'calls_0', 31, 9, 'ANSWERED', 3, '', '212996', '')] [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:22] VERBOSE[14802] chan_sip.c: Audio is at 17200 [Aug 18 10:34:22] VERBOSE[14802] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:22] VERBOSE[14802] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:22] VERBOSE[14802] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (4) INVITE - 5 [Aug 18 10:34:22] DEBUG[14799] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTP changing ssrc from 1605680423 to 1317059223 due to a source change [Aug 18 10:34:22] DEBUG[12959] stasis/app.c: Bridge 'c8381fea-1239-48c9-a6e3-1d9ad7226cf1' is 2 interested in calls_0 [Aug 18 10:34:22] DEBUG[14803] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14803] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:22] DEBUG[14804] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14793] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:22] DEBUG[14793] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[14803] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14804] http.c: HTTP Request URI is /ari/bridges/c8381fea-1239-48c9-a6e3-1d9ad7226cf1/record?name=212980_dquqCSdGIzjcjMVjdrEqzTurcKoxWjar&format=wav [Aug 18 10:34:22] DEBUG[14803] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14803] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14803] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Finding handler for bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Finding handler for bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:22] DEBUG[14803] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:22] DEBUG[14803] stasis.c: Creating topic. name: bridge:f58763a3-c201-4609-b9b6-f8cb14b257ad, detail: [Aug 18 10:34:22] DEBUG[14803] stasis.c: Topic 'bridge:f58763a3-c201-4609-b9b6-f8cb14b257ad': 0x7f0ca8004160 created [Aug 18 10:34:22] DEBUG[14803] stasis.c: Creating topic. name: cache:563/bridge:f58763a3-c201-4609-b9b6-f8cb14b257ad, detail: [Aug 18 10:34:22] DEBUG[14803] stasis.c: Topic 'cache:563/bridge:f58763a3-c201-4609-b9b6-f8cb14b257ad': 0x7f0ca8003f90 created [Aug 18 10:34:22] DEBUG[14803] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' can not use native RTP bridge as two channels are required [Aug 18 10:34:22] DEBUG[14803] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:22] DEBUG[14803] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:22] DEBUG[14803] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:22] DEBUG[14803] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:22] DEBUG[14803] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: calling simple_bridge technology constructor [Aug 18 10:34:22] DEBUG[14803] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: calling simple_bridge technology start [Aug 18 10:34:22] DEBUG[14804] http.c: match request [ari/bridges/c8381fea-1239-48c9-a6e3-1d9ad7226cf1/record] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Initializing initreq for method INVITE - callid 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Destroying SIP dialog 747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116893@178.62.121.41 SIP/2.0 [Aug 18 10:34:22] DEBUG[14805] http.c: HTTP opening session. Top level [Aug 18 10:34:22] DEBUG[14803] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:22] DEBUG[14803] http.c: HTTP closing session. Top level [Aug 18 10:34:22] DEBUG[14804] http.c: match request [ari/bridges/c8381fea-1239-48c9-a6e3-1d9ad7226cf1/record] with handler [phoneprov] len 9 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '747ca6e30003358d2e953f7b1aa3078d@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:22] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282862.490, detail: [Aug 18 10:34:22] DEBUG[14788] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:22] DEBUG[14804] http.c: match request [ari/bridges/c8381fea-1239-48c9-a6e3-1d9ad7226cf1/record] with handler [ari] len 3 [Aug 18 10:34:22] DEBUG[14804] http.c: Match made with [ari] [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'channel:1629282862.490': 0x7f0c300e4c40 created [Aug 18 10:34:22] DEBUG[20545] stasis.c: Creating topic. name: cache:564/channel:1629282862.490, detail: [Aug 18 10:34:22] DEBUG[20545] stasis.c: Topic 'cache:564/channel:1629282862.490': 0x7f0c30162ed0 created [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Finding handler for bridges/c8381fea-1239-48c9-a6e3-1d9ad7226cf1/record [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Finding handler for bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:22] DEBUG[14805] http.c: HTTP Request URI is /ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/addChannel?channel=213016 [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Finding handler for c8381fea-1239-48c9-a6e3-1d9ad7226cf1 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84126bb0) DTLS stop [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84126bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84126bb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84126bb0) ICE RTP transport deallocating [Aug 18 10:34:22] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c84126bb0' [Aug 18 10:34:22] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060 [Aug 18 10:34:22] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5c8f415c381f4c94781af7de746592ae@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38078a20) DTLS stop [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38078a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38078a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:22] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c38078a20) ICE RTP transport deallocating [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 3 [ 52]: From: ;tag=as2846b9c9 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 6 [ 60]: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:22 GMT [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:22] VERBOSE[14802] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #86 [Aug 18 10:34:22] DEBUG[14802] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:22] DEBUG[14788] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:22] DEBUG[14804] res_ari.c: No explicit handler found for c8381fea-1239-48c9-a6e3-1d9ad7226cf1. Using wildcard bridgeId. [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:564/channel:1629282862.490, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:564/channel:1629282862.490': 0x7f0c30162ed0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282862.490, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282862.490': 0x7f0c300e4c40 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:12', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000008d', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213104', '')] [Aug 18 10:34:23] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE add candidate: 10.131.0.10:14666, 2130706431 [Aug 18 10:34:23] DEBUG[14788] rtp_engine.c: RTP instance '0x7f0c3c159c70' is setup and ready to go [Aug 18 10:34:23] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) ICE stopped [Aug 18 10:34:23] DEBUG[14788] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:23] DEBUG[14788] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:22] DEBUG[14805] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:22] DEBUG[14804] res_ari.c: Finding handler for record [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c38078a20' [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:23] DEBUG[14804] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:23] DEBUG[14527] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[14527] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14805] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14805] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/addChannel] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14805] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Finding handler for bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/addChannel [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Finding handler for f58763a3-c201-4609-b9b6-f8cb14b257ad [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14805] res_ari.c: No explicit handler found for f58763a3-c201-4609-b9b6-f8cb14b257ad. Using wildcard bridgeId. [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Finding handler for addChannel [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.491, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.491': 0x7f0c30162ed0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:565/channel:1629282863.491, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:565/channel:1629282863.491': 0x7f0c300e4c40 created [Aug 18 10:34:23] DEBUG[14805] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Session timer started: 95 - 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 1768000ms [Aug 18 10:34:23] VERBOSE[14802] dial.c: Called zvonobot/79821116893 [Aug 18 10:34:23] DEBUG[14805] stasis/control.c: 213016: Sending channel add_to_bridge command [Aug 18 10:34:23] DEBUG[14788] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:23] DEBUG[14788] res_rtp_asterisk.c: (0x7f0c3c159c70) RTCP setup on RTP instance [Aug 18 10:34:23] VERBOSE[14788] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:23] DEBUG[14788] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:23] DEBUG[14788] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:23] DEBUG[14788] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:23] DEBUG[14788] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14788] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:565/channel:1629282863.491, detail: [Aug 18 10:34:23] DEBUG[13587] res_rtp_asterisk.c: (0x7f0c100e4670) RTP 0x7f0c100e6190 -- Received packet from 178.62.121.41:19902, dropping due to strict RTP protection. [Aug 18 10:34:23] DEBUG[13425] res_rtp_asterisk.c: (0x7f0c40036750) RTP 0x7f0c40052640 -- Received packet from 178.62.121.41:10972, dropping due to strict RTP protection. [Aug 18 10:34:23] DEBUG[14804] stasis.c: Creating topic. name: channel:1629282863.492, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:565/channel:1629282863.491': 0x7f0c300e4c40 destroyed [Aug 18 10:34:23] DEBUG[14386] chan_sip.c: Hangup call SIP/zvonobot-00000091, SIP callid 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14386] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:23] DEBUG[14386] res_rtp_asterisk.c: (0x7f0c80061fd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14386] res_rtp_asterisk.c: (0x7f0c80061fd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14386] channel.c: Channel 0x7f0c80074680 'SIP/zvonobot-00000091' destroying [Aug 18 10:34:23] DEBUG[14804] stasis.c: Topic 'channel:1629282863.492': 0x7f0c9c038cd0 created [Aug 18 10:34:23] DEBUG[14524] channel.c: Channel 0x7f0c7c013ef0 'Recorder/ARI-0000003b;2' allocated [Aug 18 10:34:23] DEBUG[14524] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.491, detail: [Aug 18 10:34:23] DEBUG[14528] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.491': 0x7f0c30162ed0 destroyed [Aug 18 10:34:23] DEBUG[14528] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213111': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213111' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[14386] stasis.c: Destroying topic. name: cache:352/channel:213111, detail: [Aug 18 10:34:23] DEBUG[14386] stasis.c: Topic 'cache:352/channel:213111': 0x7f0c800317c0 destroyed [Aug 18 10:34:23] DEBUG[14386] stasis.c: Destroying topic. name: channel:213111, detail: [Aug 18 10:34:23] DEBUG[14386] stasis.c: Topic 'channel:213111': 0x7f0c80030290 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:12', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000008c', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213106', '')] [Aug 18 10:34:23] DEBUG[14808] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14808] http.c: HTTP Request URI is /ari/channels/213111 [Aug 18 10:34:23] DEBUG[14804] stasis.c: Creating topic. name: cache:566/channel:1629282863.492, detail: [Aug 18 10:34:23] DEBUG[14807] bridge_channel.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: 0x7f0c7c02d4b0(Recorder/ARI-0000003b;2) is joining [Aug 18 10:34:23] DEBUG[14808] http.c: match request [ari/channels/213111] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000002b - start 1629282830.096747 answer 1629282836.147689 end 1629282862.725575 dur 32.628 bill 26.577 dispo ANSWERED [Aug 18 10:34:23] DEBUG[14808] http.c: match request [ari/channels/213111] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14788] chan_sip.c: SIP call-id changed from '66bea07450b6898d59e5b9a461a6d094@127.0.1.1:5060' to '25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060' [Aug 18 10:34:23] DEBUG[14808] http.c: match request [ari/channels/213111] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14804] stasis.c: Topic 'cache:566/channel:1629282863.492': 0x7f0c9c053730 created [Aug 18 10:34:23] DEBUG[14808] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14788] stasis.c: Creating topic. name: channel:213172, detail: [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 From: ;tag=as2d7c4d21 To: ;tag=as6fcc16b3 Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[14652] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0' [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.494, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[14319] bridge_channel.c: Setting 0x7f0c180c8c80(Recorder/ARI-00000030;2) state from:0 to:1 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.494': 0x7f0c300e4c40 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:567/channel:1629282863.494, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:567/channel:1629282863.494': 0x7f0c30162ed0 created [Aug 18 10:34:23] DEBUG[14652] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:23] DEBUG[14652] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[13394] bridge_channel.c: Setting 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) state from:0 to:1 [Aug 18 10:34:23] DEBUG[14809] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14331] channel.c: Channel 0x7f0c180999a0 'Recorder/ARI-00000030;1' destroying [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Finding handler for channels/213111 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14319] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: pulling 0x7f0c180c8c80(Recorder/ARI-00000030;2) [Aug 18 10:34:23] DEBUG[14807] bridge_channel.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: pushing 0x7f0c7c02d4b0(Recorder/ARI-0000003b;2) [Aug 18 10:34:23] DEBUG[14788] stasis.c: Topic 'channel:213172': 0x7f0c3c08e200 created [Aug 18 10:34:23] VERBOSE[14319] bridge_channel.c: Channel Recorder/ARI-00000030;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:23] DEBUG[14788] stasis.c: Creating topic. name: cache:568/channel:213172, detail: [Aug 18 10:34:23] DEBUG[14788] stasis.c: Topic 'cache:568/channel:213172': 0x7f0c3c13f0d0 created [Aug 18 10:34:23] DEBUG[14319] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c180c8c80(Recorder/ARI-00000030;2) is leaving simple_bridge technology [Aug 18 10:34:23] DEBUG[13394] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: pulling 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) [Aug 18 10:34:23] DEBUG[14809] http.c: HTTP Request URI is /ari/channels/212972 [Aug 18 10:34:23] VERBOSE[13394] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 left 'simple_bridge' stasis-bridge <4918ac35-38b0-4486-b626-7cf67dacf45b> [Aug 18 10:34:23] DEBUG[14319] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[13394] bridge_channel.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) is leaving simple_bridge technology [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d7c4d21 [Aug 18 10:34:23] DEBUG[14331] stasis.c: Destroying topic. name: cache:320/channel:1629282848.273, detail: [Aug 18 10:34:23] DEBUG[14331] stasis.c: Topic 'cache:320/channel:1629282848.273': 0x7f0c180bfd90 destroyed [Aug 18 10:34:23] DEBUG[14331] stasis.c: Destroying topic. name: channel:1629282848.273, detail: [Aug 18 10:34:23] DEBUG[14331] stasis.c: Topic 'channel:1629282848.273': 0x7f0c180bacb0 destroyed [Aug 18 10:34:23] DEBUG[14319] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[13394] bridge_native_rtp.c: Bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[13394] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[13394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[13394] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[13394] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[13394] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b is already using the new technology. [Aug 18 10:34:23] DEBUG[13394] bridge_channel.c: Bridge is returning 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) to write format slin16 [Aug 18 10:34:23] DEBUG[13394] channel.c: Channel UnicastRTP/127.0.0.1:50283-0x7f0c840529d0 setting write format path: slin16 -> slin16 [Aug 18 10:34:23] DEBUG[13394] stasis/control.c: robot_212972, 4918ac35-38b0-4486-b626-7cf67dacf45b: Channel was departed from bridge [Aug 18 10:34:23] DEBUG[13394] stasis/app.c: bridge '4918ac35-38b0-4486-b626-7cf67dacf45b': is 2 interested in calls_0 [Aug 18 10:34:23] DEBUG[13394] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:23] DEBUG[14319] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[13377] stasis/control.c: robot_212972: Channel departing bridge [Aug 18 10:34:23] DEBUG[14319] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14809] http.c: match request [ari/channels/212972] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[13377] bridge.c: Waiting for 0x7f0ca803b9d0(UnicastRTP/127.0.0.1:50283-0x7f0c840529d0) bridge thread to die. [Aug 18 10:34:23] DEBUG[13377] stasis/app.c: channel 'robot_212972': is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[13377] channel.c: Channel 0x7f0c84060000 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[14809] http.c: match request [ari/channels/212972] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6fcc16b3 [Aug 18 10:34:23] DEBUG[14809] http.c: match request [ari/channels/212972] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14319] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14319] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:23] DEBUG[14319] channel.c: Channel 0x7f0c180f2f90 'Recorder/ARI-00000030;2' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:567/channel:1629282863.494, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:567/channel:1629282863.494': 0x7f0c30162ed0 destroyed [Aug 18 10:34:23] DEBUG[14809] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14807] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:23] VERBOSE[14807] bridge_channel.c: Channel Recorder/ARI-0000003b;2 joined 'simple_bridge' stasis-bridge <26acc09b-99c1-4bbb-afbd-344c8a9a505d> [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.494, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.494': 0x7f0c300e4c40 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000008f', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213107', '')] [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Finding handler for channels/212972 [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Finding handler for 212972 [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking channels create: Didn't match 212972 [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14809] res_ari.c: Checking channels externalMedia: Didn't match 212972 [Aug 18 10:34:23] DEBUG[14809] res_ari.c: No explicit handler found for 212972. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Finding handler for 213111 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking channels create: Didn't match 213111 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14808] res_ari.c: Checking channels externalMedia: Didn't match 213111 [Aug 18 10:34:23] DEBUG[14808] res_ari.c: No explicit handler found for 213111. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] DEBUG[14807] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d'. Checking compatability for channels 'SIP/zvonobot-00000049' and 'Recorder/ARI-0000003b;2' [Aug 18 10:34:23] DEBUG[14807] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' can not use native RTP bridge as could not get details [Aug 18 10:34:23] DEBUG[14807] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14807] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14807] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14807] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14807] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d is already using the new technology. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[14807] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d: 0x7f0c7c02d4b0(Recorder/ARI-0000003b;2) is joining simple_bridge technology [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 (Checking To) --From tag as2d7c4d21 --To-tag as6fcc16b3 [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel Recorder/ARI-0000003b;2 setting read format path: slin -> slin [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[13271] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000034 [Aug 18 10:34:23] DEBUG[13271] stasis/control.c: 213016: Adding to bridge f58763a3-c201-4609-b9b6-f8cb14b257ad [Aug 18 10:34:23] DEBUG[13271] stasis/app.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel SIP/zvonobot-00000049 setting write format path: slin -> ulaw [Aug 18 10:34:23] DEBUG[14530] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #129 (5) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #129)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116897@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9 Max-Forwards: 70 From: ;tag=as5acf84f3 To: Contact: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1168069357 1168069357 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18198 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14810] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c3017b4a0(SIP/zvonobot-00000034) is joining [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14530] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.495, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (3) INVITE - 5 [Aug 18 10:34:23] DEBUG[14669] stasis.c: Creating topic. name: channel:1629282863.496, detail: [Aug 18 10:34:23] DEBUG[14669] stasis.c: Topic 'channel:1629282863.496': 0x7f0c8c03a8f0 created [Aug 18 10:34:23] DEBUG[14669] stasis.c: Creating topic. name: cache:569/channel:1629282863.496, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.495': 0x7f0c30162ed0 created [Aug 18 10:34:23] DEBUG[14669] stasis.c: Topic 'cache:569/channel:1629282863.496': 0x7f0c8c044af0 created [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:570/channel:1629282863.495, detail: [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116889@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 Max-Forwards: 70 From: ;tag=as1ac06673 To: Contact: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 722698992 722698992 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel SIP/zvonobot-00000049 setting read format path: ulaw -> slin [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #7 (5) INVITE - 5 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:570/channel:1629282863.495': 0x7f0c300e4c40 created [Aug 18 10:34:23] DEBUG[14541] channel.c: Channel 0x7f0cb40970e0 'Recorder/ARI-0000003c;2' allocated [Aug 18 10:34:23] DEBUG[14541] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:23] DEBUG[14395] chan_sip.c: Hangup call SIP/zvonobot-00000092, SIP callid 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14395] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:23] DEBUG[14395] res_rtp_asterisk.c: (0x7f0c8c104f10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14395] res_rtp_asterisk.c: (0x7f0c8c104f10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14395] channel.c: Channel 0x7f0c8c11cb60 'SIP/zvonobot-00000092' destroying [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213112': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213112' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[14395] stasis.c: Destroying topic. name: cache:355/channel:213112, detail: [Aug 18 10:34:23] DEBUG[14395] stasis.c: Topic 'cache:355/channel:213112': 0x7f0c8c11ffc0 destroyed [Aug 18 10:34:23] DEBUG[14395] stasis.c: Destroying topic. name: channel:213112, detail: [Aug 18 10:34:23] DEBUG[14395] stasis.c: Topic 'channel:213112': 0x7f0c8c11e8c0 destroyed [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #7)) [Aug 18 10:34:23] DEBUG[14811] bridge_channel.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: 0x7f0cb4010fa0(Recorder/ARI-0000003c;2) is joining [Aug 18 10:34:23] DEBUG[14454] channel.c: Channel 0x7f0c880272f0 'SIP/zvonobot-0000001c' destroying [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116901@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b Max-Forwards: 70 From: ;tag=as3a1d6e7b To: Contact: Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1147590624 1147590624 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14536] stasis.c: Creating topic. name: channel:1629282863.497, detail: [Aug 18 10:34:23] DEBUG[14536] stasis.c: Topic 'channel:1629282863.497': 0x7f0ca4111770 created [Aug 18 10:34:23] DEBUG[14536] stasis.c: Creating topic. name: cache:571/channel:1629282863.497, detail: [Aug 18 10:34:23] DEBUG[14536] stasis.c: Topic 'cache:571/channel:1629282863.497': 0x7f0ca40f9fe0 created [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '212991': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '212991' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel Recorder/ARI-0000003b;2 setting write format path: slin -> slin [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Destroying SIP dialog 217865353a22dc3331285fc05bb15812@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '217865353a22dc3331285fc05bb15812@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c80061fd0) DTLS stop [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c80061fd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c80061fd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c80061fd0) ICE RTP transport deallocating [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c80061fd0' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (1) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1216361947 1216361947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 From: ;tag=as3a3fa466 To: ;tag=as28515388 Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a3fa466 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as28515388 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 (Checking To) --From tag as3a3fa466 --To-tag as28515388 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Destroying SIP dialog 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '07f82ab44968293544eb273a476d91c1@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c104f10) DTLS stop [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c104f10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c104f10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c104f10) ICE RTP transport deallocating [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8c104f10' [Aug 18 10:34:23] DEBUG[14812] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14810] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: pushing 0x7f0c3017b4a0(SIP/zvonobot-00000034) [Aug 18 10:34:23] VERBOSE[14810] bridge_channel.c: Channel SIP/zvonobot-00000034 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:23] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:23] DEBUG[14813] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[13627] res_rtp_asterisk.c: (0x7f0c7c01e650) RTCP got report of 100 bytes from 178.62.121.41:10225 [Aug 18 10:34:23] DEBUG[14812] http.c: HTTP Request URI is /ari/channels/213112 [Aug 18 10:34:23] DEBUG[14454] stasis.c: Destroying topic. name: cache:35/channel:212991, detail: [Aug 18 10:34:23] DEBUG[14810] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[14454] stasis.c: Topic 'cache:35/channel:212991': 0x7f0c88029ab0 destroyed [Aug 18 10:34:23] DEBUG[14813] http.c: HTTP Request URI is /ari/channels/213015/snoop?app=calls_0&spy=in [Aug 18 10:34:23] DEBUG[14454] stasis.c: Destroying topic. name: channel:212991, detail: [Aug 18 10:34:23] DEBUG[14454] stasis.c: Topic 'channel:212991': 0x7f0c8802ad00 destroyed [Aug 18 10:34:23] DEBUG[14454] channel.c: Channel 0x7f0c7008d490 'Snoop/212991-00000011' destroying [Aug 18 10:34:23] DEBUG[14394] res_rtp_asterisk.c: (0x7f0c38023850) RTCP got report of 76 bytes from 178.62.121.41:12659 [Aug 18 10:34:23] DEBUG[14812] http.c: match request [ari/channels/213112] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14813] http.c: match request [ari/channels/213015/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14810] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14812] http.c: match request [ari/channels/213112] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14813] http.c: match request [ari/channels/213015/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14454] stasis.c: Destroying topic. name: cache:371/channel:1629282851.322, detail: [Aug 18 10:34:23] DEBUG[14454] stasis.c: Topic 'cache:371/channel:1629282851.322': 0x7f0c700a14f0 destroyed [Aug 18 10:34:23] DEBUG[14810] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14813] http.c: match request [ari/channels/213015/snoop] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14813] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14812] http.c: match request [ari/channels/213112] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:570/channel:1629282863.495, detail: [Aug 18 10:34:23] DEBUG[14811] bridge_channel.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: pushing 0x7f0cb4010fa0(Recorder/ARI-0000003c;2) [Aug 18 10:34:23] DEBUG[14810] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Finding handler for channels/213015/snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Finding handler for 213015 [Aug 18 10:34:23] DEBUG[14454] stasis.c: Destroying topic. name: channel:1629282851.322, detail: [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channels create: Didn't match 213015 [Aug 18 10:34:23] DEBUG[14454] stasis.c: Topic 'channel:1629282851.322': 0x7f0c700a3070 destroyed [Aug 18 10:34:23] DEBUG[14810] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:570/channel:1629282863.495': 0x7f0c300e4c40 destroyed [Aug 18 10:34:23] DEBUG[14810] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad is already using the new technology. [Aug 18 10:34:23] DEBUG[14812] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14810] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c3017b4a0(SIP/zvonobot-00000034) is joining simple_bridge technology [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channels externalMedia: Didn't match 213015 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.495, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.495': 0x7f0c30162ed0 destroyed [Aug 18 10:34:23] DEBUG[14813] res_ari.c: No explicit handler found for 213015. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Finding handler for channels/213112 [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:12', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000008e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213108', '')] [Aug 18 10:34:23] DEBUG[14810] res_rtp_asterisk.c: (0x7f0c20028ba0) RTP changing ssrc from 442010798 to 1445110902 due to a source change [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Finding handler for snoop [Aug 18 10:34:23] DEBUG[13271] stasis/app.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' is 2 interested in calls_0 [Aug 18 10:34:23] DEBUG[14805] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14805] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 From: ;tag=as2ed109a6 To: ;tag=as7faa24c6 Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:23] DEBUG[14524] res_stasis_recording.c: 1629282856.390: Sending record(213038_BKsPPDtGadkoThbYiyRxVjYHUBQlBmjs.wav) command [Aug 18 10:34:23] DEBUG[14811] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Finding handler for 213112 [Aug 18 10:34:23] VERBOSE[14811] bridge_channel.c: Channel Recorder/ARI-0000003c;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:23] DEBUG[14396] chan_sip.c: Hangup call SIP/zvonobot-00000093, SIP callid 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14393] chan_sip.c: Hangup call SIP/zvonobot-00000090, SIP callid 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212972-00000006 - start 1629282833.573782 answer 1629282833.573782 end 1629282862.813726 dur 29.239 bill 29.239 dispo ANSWERED [Aug 18 10:34:23] DEBUG[14396] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:23] DEBUG[14396] res_rtp_asterisk.c: (0x7f0c90058ee0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14524] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:23] DEBUG[14524] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14396] res_rtp_asterisk.c: (0x7f0c90058ee0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14396] channel.c: Channel 0x7f0c900a0ad0 'SIP/zvonobot-00000093' destroying [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking channels create: Didn't match 213112 [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14812] res_ari.c: Checking channels externalMedia: Didn't match 213112 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[14814] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14679] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[14679] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14393] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ed109a6 [Aug 18 10:34:23] DEBUG[14393] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14393] channel.c: Channel 0x7f0c78053fb0 'SIP/zvonobot-00000090' destroying [Aug 18 10:34:23] DEBUG[14815] app.c: play_and_record: , /var/spool/asterisk/recording/213038_BKsPPDtGadkoThbYiyRxVjYHUBQlBmjs, 'wav' [Aug 18 10:34:23] DEBUG[14815] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:23] VERBOSE[14815] app.c: x=0, open writing: /var/spool/asterisk/recording/213038_BKsPPDtGadkoThbYiyRxVjYHUBQlBmjs format: wav, 0x7f0c1c1499f0 [Aug 18 10:34:23] DEBUG[14816] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14814] http.c: HTTP Request URI is /ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/record?name=213016_MtjOSnOVSDieYyOzUQyEuUjYfOSqPhfe&format=wav [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:23] DEBUG[14812] res_ari.c: No explicit handler found for 213112. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14816] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:23] DEBUG[14814] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/record] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213113': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213113' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis.c: Destroying topic. name: cache:356/channel:213113, detail: [Aug 18 10:34:23] DEBUG[20620] stasis.c: Topic 'cache:356/channel:213113': 0x7f0c9007e730 destroyed [Aug 18 10:34:23] DEBUG[20620] stasis.c: Destroying topic. name: channel:213113, detail: [Aug 18 10:34:23] DEBUG[20620] stasis.c: Topic 'channel:213113': 0x7f0c900666d0 destroyed [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213109': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213109' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[14819] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14814] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/record] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:23] DEBUG[14813] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:23] DEBUG[14819] http.c: HTTP Request URI is /ari/channels/213113 [Aug 18 10:34:23] DEBUG[14393] stasis.c: Destroying topic. name: cache:353/channel:213109, detail: [Aug 18 10:34:23] DEBUG[14814] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/record] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14820] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14393] stasis.c: Topic 'cache:353/channel:213109': 0x7f0c7802e7f0 destroyed [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7faa24c6 [Aug 18 10:34:23] DEBUG[14814] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14811] bridge_native_rtp.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e'. Checking compatability for channels 'SIP/zvonobot-00000026' and 'Recorder/ARI-0000003c;2' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14816] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14811] bridge_native_rtp.c: Bridge 'cd02ec13-331f-440c-a360-837dbdfdba5e' can not use native RTP bridge as could not get details [Aug 18 10:34:23] DEBUG[14819] http.c: match request [ari/channels/213113] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.498, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14820] http.c: HTTP Request URI is /ari/channels/213109 [Aug 18 10:34:23] DEBUG[14393] stasis.c: Destroying topic. name: channel:213109, detail: [Aug 18 10:34:23] DEBUG[14811] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.498': 0x7f0c30162ed0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:572/channel:1629282863.498, detail: [Aug 18 10:34:23] DEBUG[14393] stasis.c: Topic 'channel:213109': 0x7f0c7807b2f0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:572/channel:1629282863.498': 0x7f0c3005c090 created [Aug 18 10:34:23] DEBUG[14820] http.c: match request [ari/channels/213109] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Finding handler for bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/record [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14811] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14816] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14820] http.c: match request [ari/channels/213109] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14819] http.c: match request [ari/channels/213113] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] DEBUG[14819] http.c: match request [ari/channels/213113] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14819] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[14811] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14811] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14820] http.c: match request [ari/channels/213109] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:572/channel:1629282863.498, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:572/channel:1629282863.498': 0x7f0c3005c090 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.498, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.498': 0x7f0c30162ed0 destroyed [Aug 18 10:34:23] DEBUG[14811] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e is already using the new technology. [Aug 18 10:34:23] DEBUG[14820] http.c: Match made with [ari] [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[14811] bridge.c: Bridge cd02ec13-331f-440c-a360-837dbdfdba5e: 0x7f0cb4010fa0(Recorder/ARI-0000003c;2) is joining simple_bridge technology [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel Recorder/ARI-0000003c;2 setting read format path: slin -> slin [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel SIP/zvonobot-00000026 setting write format path: slin -> ulaw [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel SIP/zvonobot-00000026 setting read format path: ulaw -> slin [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel Recorder/ARI-0000003c;2 setting write format path: slin -> slin [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Finding handler for channels/213113 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000091', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213111', '')] [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Finding handler for channels/213109 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 (Checking To) --From tag as2ed109a6 --To-tag as7faa24c6 [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14816] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14816] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14436] channel.c: Channel 0x7f0cac056530 'SIP/zvonobot-000000a6' allocated [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14436] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.499, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.499': 0x7f0c300e4c40 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:573/channel:1629282863.499, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:573/channel:1629282863.499': 0x7f0c3017d6c0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:573/channel:1629282863.499, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:573/channel:1629282863.499': 0x7f0c3017d6c0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.499, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.499': 0x7f0c300e4c40 destroyed [Aug 18 10:34:23] DEBUG[14436] res_stasis.c: calls_0: Subscribing to 213131 [Aug 18 10:34:23] DEBUG[14436] stasis/app.c: Channel '213131' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Finding handler for f58763a3-c201-4609-b9b6-f8cb14b257ad [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14814] res_ari.c: No explicit handler found for f58763a3-c201-4609-b9b6-f8cb14b257ad. Using wildcard bridgeId. [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Finding handler for record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:23] DEBUG[14814] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000092', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213112', '')] [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel Recorder/ARI-0000003b;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14814] stasis.c: Creating topic. name: channel:1629282863.500, detail: [Aug 18 10:34:23] DEBUG[14807] channel.c: Channel Recorder/ARI-0000003b;2 setting write format path: alaw -> slin [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Finding handler for 213113 [Aug 18 10:34:23] DEBUG[14436] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14436] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2052644047 2052644047 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16832 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking channels create: Didn't match 213113 [Aug 18 10:34:23] DEBUG[14814] stasis.c: Topic 'channel:1629282863.500': 0x7f0c0806ec20 created [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Finding handler for 213109 [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Outgoing Call for 79821116909 [Aug 18 10:34:23] DEBUG[14814] stasis.c: Creating topic. name: cache:574/channel:1629282863.500, detail: [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking channels create: Didn't match 213109 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: Checking channels externalMedia: Didn't match 213113 [Aug 18 10:34:23] DEBUG[14709] channel.c: Channel 0x7f0c7c0e0450 'Recorder/ARI-00000045;1' allocated [Aug 18 10:34:23] DEBUG[14709] stasis.c: Creating topic. name: channel:1629282863.501, detail: [Aug 18 10:34:23] DEBUG[14709] stasis.c: Topic 'channel:1629282863.501': 0x7f0c7c08ce80 created [Aug 18 10:34:23] DEBUG[14709] stasis.c: Creating topic. name: cache:575/channel:1629282863.501, detail: [Aug 18 10:34:23] DEBUG[14709] stasis.c: Topic 'cache:575/channel:1629282863.501': 0x7f0c7c009b20 created [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6348ms with no response [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Hanging up call 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14690] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (2) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116894@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051 Max-Forwards: 70 From: ;tag=as3e829f44 To: Contact: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1459261102 1459261102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6422ms with no response [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Hanging up call 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14535] channel.c: Channel 0x7f0ca01129e0 'SIP/zvonobot-000000a1' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.502, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.502': 0x7f0c3017d6c0 created [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:576/channel:1629282863.502, detail: [Aug 18 10:34:23] DEBUG[14690] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14534] channel.c: Channel 0x7f0c8009df10 'SIP/zvonobot-000000a0' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[14820] res_ari.c: Checking channels externalMedia: Didn't match 213109 [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6217ms with no response [Aug 18 10:34:23] DEBUG[14814] stasis.c: Topic 'cache:574/channel:1629282863.500': 0x7f0c080767f0 created [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Hanging up call 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:576/channel:1629282863.502': 0x7f0c3017d7a0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:576/channel:1629282863.502, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:576/channel:1629282863.502': 0x7f0c3017d7a0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.502, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.502': 0x7f0c3017d6c0 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000001c', '', 'Stasis', 'calls_0', 31, 13, 'ANSWERED', 3, '', '212991', '')] [Aug 18 10:34:23] DEBUG[14820] res_ari.c: No explicit handler found for 213109. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14540] channel.c: Channel 0x7f0c9c0d6ff0 'SIP/zvonobot-000000a2' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[14541] res_stasis_recording.c: 1629282857.398: Sending record(213003_eusdgnrhynNdoWptPwesQBAJOQYsAQjr.wav) command [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218;received=159.65.48.104 From: ;tag=as41c9ab07 To: ;tag=as255cc2fc Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31a18a3d" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as41c9ab07 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as255cc2fc [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31a18a3d" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 (Checking To) --From tag as41c9ab07 --To-tag as255cc2fc [Aug 18 10:34:23] DEBUG[14824] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14824] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:23] DEBUG[14824] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14541] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:23] DEBUG[14819] res_ari.c: No explicit handler found for 213113. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14824] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Stopping retransmission on '5129740f51f9292d29e823f263748e28@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116921@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327f6218 Max-Forwards: 70 From: ;tag=as41c9ab07 To: Contact: Call-ID: 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14824] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14824] http.c: Match made with [ari] [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6167ms with no response [Aug 18 10:34:23] WARNING[20585] chan_sip.c: Hanging up call 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14541] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14542] channel.c: Channel 0x7f0ca4122130 'SIP/zvonobot-000000a3' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Destroying SIP dialog 70757ec224866cc54887d48e040f5301@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:23] VERBOSE[14821] chan_sip.c: Audio is at 10524 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '70757ec224866cc54887d48e040f5301@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] VERBOSE[14821] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212991-00000011 - start 1629282854.752415 answer 1629282854.752415 end 1629282863.264507 dur 8.512 bill 8.512 dispo ANSWERED [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90058ee0) DTLS stop [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90058ee0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90058ee0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c90058ee0) ICE RTP transport deallocating [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c90058ee0' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Destroying SIP dialog 577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '577acf1c2f97566d127d02eb3be5f520@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS stop [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7806cff0) ICE RTP transport deallocating [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7806cff0' [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245;received=159.65.48.104 From: ;tag=as3ee6d51f To: ;tag=as33cfbf7c Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ccd2abc" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK58fea245;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3ee6d51f [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as33cfbf7c [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ccd2abc" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 (Checking To) --From tag as3ee6d51f --To-tag as33cfbf7c [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1bf217940e05c52a3668e71801db4d97@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116892@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 Max-Forwards: 70 From: ;tag=as0a953bb4 To: Contact: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 772240936 772240936 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[14823] app.c: play_and_record: , /var/spool/asterisk/recording/213003_eusdgnrhynNdoWptPwesQBAJOQYsAQjr, 'wav' [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.503, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.503': 0x7f0c3017d7a0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:577/channel:1629282863.503, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:577/channel:1629282863.503': 0x7f0c3017d6c0 created [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14823] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:23] DEBUG[14816] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] VERBOSE[14821] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] DEBUG[14816] stasis.c: Creating topic. name: bridge:d9991214-2fd4-4dac-a3cc-4af033e7cd5f, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:577/channel:1629282863.503, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:577/channel:1629282863.503': 0x7f0c3017d6c0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.503, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.503': 0x7f0c3017d7a0 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:14', '"" <>', '', 's', 'default', 'Snoop/212991-00000011', '', 'Stasis', 'calls_0', 8, 8, 'ANSWERED', 3, '', '1629282851.322', '')] [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[14824] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] DEBUG[14824] stasis.c: Creating topic. name: bridge:d8303a18-9287-4546-8778-524203bcb0a2, detail: [Aug 18 10:34:23] DEBUG[14824] stasis.c: Topic 'bridge:d8303a18-9287-4546-8778-524203bcb0a2': 0x7f0c34076d30 created [Aug 18 10:34:23] DEBUG[14824] stasis.c: Creating topic. name: cache:578/bridge:d8303a18-9287-4546-8778-524203bcb0a2, detail: [Aug 18 10:34:23] DEBUG[14824] stasis.c: Topic 'cache:578/bridge:d8303a18-9287-4546-8778-524203bcb0a2': 0x7f0c3401cf30 created [Aug 18 10:34:23] VERBOSE[14821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43;received=159.65.48.104 From: ;tag=as126e0733 To: ;tag=as7887a81a Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b521c7e" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel Recorder/ARI-0000003c;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:23] DEBUG[14811] channel.c: Channel Recorder/ARI-0000003c;2 setting write format path: alaw -> slin [Aug 18 10:34:23] VERBOSE[14823] app.c: x=0, open writing: /var/spool/asterisk/recording/213003_eusdgnrhynNdoWptPwesQBAJOQYsAQjr format: wav, 0x7f0c280c5840 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62e18e43;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14824] bridge_native_rtp.c: Bridge 'd8303a18-9287-4546-8778-524203bcb0a2' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as126e0733 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7887a81a [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14824] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Initializing initreq for method INVITE - callid 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116909@178.62.121.41 SIP/2.0 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 3 [ 52]: From: ;tag=as3097acc5 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:23] DEBUG[14824] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14824] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:23] DEBUG[14824] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 6 [ 60]: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.504, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.504': 0x7f0c3017d6c0 created [Aug 18 10:34:23] DEBUG[14816] stasis.c: Topic 'bridge:d9991214-2fd4-4dac-a3cc-4af033e7cd5f': 0x7f0c1803cd60 created [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] VERBOSE[14821] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:34:23] DEBUG[14821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:579/channel:1629282863.504, detail: [Aug 18 10:34:23] DEBUG[14824] bridge.c: Bridge d8303a18-9287-4546-8778-524203bcb0a2: calling simple_bridge technology constructor [Aug 18 10:34:23] DEBUG[14824] bridge.c: Bridge d8303a18-9287-4546-8778-524203bcb0a2: calling simple_bridge technology start [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:579/channel:1629282863.504': 0x7f0c3002e830 created [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b521c7e" [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 137837c51322c444587a45b5059337ee@159.65.48.104:5060 (Checking To) --From tag as126e0733 --To-tag as7887a81a [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 137837c51322c444587a45b5059337ee@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 From: ;tag=as000dc064 To: ;tag=as67a63ec0 Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62799cef;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[14816] stasis.c: Creating topic. name: cache:580/bridge:d9991214-2fd4-4dac-a3cc-4af033e7cd5f, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as000dc064 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as67a63ec0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14824] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14824] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14825] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14825] http.c: HTTP Request URI is /ari/channels/213003/snoop?app=calls_0&spy=in [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:579/channel:1629282863.504, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:579/channel:1629282863.504': 0x7f0c3002e830 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.504, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.504': 0x7f0c3017d6c0 destroyed [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000093', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213113', '')] [Aug 18 10:34:23] DEBUG[14688] channel.c: Channel 0x7f0c18029cf0 'SIP/zvonobot-000000be' allocated [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14688] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16a97228" [Aug 18 10:34:23] DEBUG[14816] stasis.c: Topic 'cache:580/bridge:d9991214-2fd4-4dac-a3cc-4af033e7cd5f': 0x7f0c1801af60 created [Aug 18 10:34:23] DEBUG[14825] http.c: match request [ari/channels/213003/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14825] http.c: match request [ari/channels/213003/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:23] VERBOSE[14821] dial.c: Called zvonobot/79821116909 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 (Checking To) --From tag as000dc064 --To-tag as67a63ec0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 78ff90082a32733c37c86b5b0b68ef9b@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4fab01364082215c146a77c82d78ca08@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4fab01364082215c146a77c82d78ca08@159.65.48.104:5060' Method: BYE [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS stop [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88003e20) ICE RTP transport deallocating [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c88003e20' [Aug 18 10:34:23] DEBUG[14825] http.c: match request [ari/channels/213003/snoop] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14688] res_stasis.c: calls_0: Subscribing to 213154 [Aug 18 10:34:23] DEBUG[14688] stasis/app.c: Channel '213154' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Outgoing Call for 79821116886 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:23] VERBOSE[14827] chan_sip.c: Audio is at 14616 [Aug 18 10:34:23] VERBOSE[14827] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:23] VERBOSE[14827] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:23] VERBOSE[14827] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:23] DEBUG[14688] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14688] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14566] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[14566] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Initializing initreq for method INVITE - callid 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116886@178.62.121.41 SIP/2.0 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 3 [ 52]: From: ;tag=as7d998899 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30440186;received=159.65.48.104 From: ;tag=as50732cd4 To: ;tag=as2fa6b009 Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 641503836 641503836 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10972 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK30440186;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as50732cd4 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2fa6b009 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 641503836 641503836 IN IP4 178.62.121.41 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10972 RTP/AVP 0 8 101 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 (Checking To) --From tag as50732cd4 --To-tag as2fa6b009 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 6 [ 60]: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14825] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Finding handler for channels/213003/snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14816] bridge_native_rtp.c: Bridge 'd9991214-2fd4-4dac-a3cc-4af033e7cd5f' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Finding handler for 213003 [Aug 18 10:34:23] DEBUG[14816] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channels create: Didn't match 213003 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:23] DEBUG[14816] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Stopping retransmission on '10f580a044264908688c62534aa40882@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channels externalMedia: Didn't match 213003 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:23] DEBUG[14825] res_ari.c: No explicit handler found for 213003. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Finding handler for snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:23] DEBUG[14816] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:23] DEBUG[14816] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14816] bridge.c: Bridge d9991214-2fd4-4dac-a3cc-4af033e7cd5f: calling simple_bridge technology constructor [Aug 18 10:34:23] DEBUG[14816] bridge.c: Bridge d9991214-2fd4-4dac-a3cc-4af033e7cd5f: calling simple_bridge technology start [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.505, detail: [Aug 18 10:34:23] DEBUG[14816] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] VERBOSE[14827] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #107 [Aug 18 10:34:23] DEBUG[14827] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Got SDP version 641503836 and unique parts [root 641503836 IN IP4 178.62.121.41] [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.505': 0x7f0c3002e830 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:581/channel:1629282863.505, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:581/channel:1629282863.505': 0x7f0c3017d6c0 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:581/channel:1629282863.505, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:581/channel:1629282863.505': 0x7f0c3017d6c0 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.505, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.505': 0x7f0c3002e830 destroyed [Aug 18 10:34:23] DEBUG[14828] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14828] http.c: HTTP Request URI is /ari/channels/213038/snoop?app=calls_0&spy=in [Aug 18 10:34:23] DEBUG[14828] http.c: match request [ari/channels/213038/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14828] http.c: match request [ari/channels/213038/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14828] http.c: match request [ari/channels/213038/snoop] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14828] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000090', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213109', '')] [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 641503836 641503836 IN IP4 178.62.121.41... OK. [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:23] DEBUG[14816] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:23] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:23] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Finding handler for channels/213038/snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Finding handler for 213038 [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channels create: Didn't match 213038 [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channels externalMedia: Didn't match 213038 [Aug 18 10:34:23] DEBUG[14828] res_ari.c: No explicit handler found for 213038. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Finding handler for snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:23] DEBUG[14828] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:23] VERBOSE[14827] dial.c: Called zvonobot/79821116886 [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:23] DEBUG[14825] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40036750) ICE set role failed; no ice instance [Aug 18 10:34:23] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40036750) RTCP setting address on RTP instance [Aug 18 10:34:23] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c40052640 -- Strict RTP learning after remote address set to: 178.62.121.41:10972 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10972 [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00d0a68) from 0x7f0c147e2330 to 0x7f0c40036928 [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb011a9c8) from 0x7f0c147e2330 to 0x7f0c40036928 [Aug 18 10:34:23] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb006a808) from 0x7f0c147e2330 to 0x7f0c40036928 [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40036750) RTCP ignoring duplicate property [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:23] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000042 setting read format path: alaw -> alaw [Aug 18 10:34:23] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000042 setting write format path: alaw -> alaw [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a1 - start 1629282857.024444 answer 0.000000 end 1629282863.476945 dur 6.452 bill 1629282863.476 dispo NO ANSWER [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a0 - start 1629282856.960044 answer 0.000000 end 1629282863.479378 dur 6.519 bill 1629282863.479 dispo NO ANSWER [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40036750) DTLS - ast_rtp_activate rtp=0x7f0c40052640 - setup and perform DTLS' [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40052640) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a2 - start 1629282857.159925 answer 0.000000 end 1629282863.497819 dur 6.337 bill 1629282863.497 dispo NO ANSWER [Aug 18 10:34:23] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40052640) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:23] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a3 - start 1629282857.203091 answer 0.000000 end 1629282863.520735 dur 6.317 bill 1629282863.520 dispo NO ANSWER [Aug 18 10:34:23] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:23] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Strict routing enforced for session 10f580a044264908688c62534aa40882@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:23] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:23] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117007@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70f552ae Max-Forwards: 70 From: ;tag=as50732cd4 To: ;tag=as2fa6b009 Contact: Call-ID: 10f580a044264908688c62534aa40882@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14564] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240' [Aug 18 10:34:23] DEBUG[14564] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:23] DEBUG[14564] http.c: HTTP closing session. Top level [Aug 18 10:34:23] VERBOSE[13425] dial.c: SIP/zvonobot-00000042 answered [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (1) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14830] http.c: HTTP opening session. Top level [Aug 18 10:34:23] VERBOSE[13425] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000042 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (3) INVITE - 5 [Aug 18 10:34:23] DEBUG[13227] bridge_channel.c: Setting 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) state from:0 to:1 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:23] DEBUG[13227] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: pulling 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) [Aug 18 10:34:23] DEBUG[13425] stasis/app.c: Channel '213033' is 2 interested in calls_0 [Aug 18 10:34:23] VERBOSE[13227] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50397-0x7f0c28011240 left 'simple_bridge' stasis-bridge <3f704757-87e2-45e5-8aa9-92ed6ea9feee> [Aug 18 10:34:23] DEBUG[13211] channel.c: Channel 0x7f0c1002e710 'Recorder/ARI-00000003;1' destroying [Aug 18 10:34:23] DEBUG[13227] bridge_channel.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) is leaving simple_bridge technology [Aug 18 10:34:23] DEBUG[14830] http.c: HTTP Request URI is /ari/channels/212983 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] VERBOSE[13425] res_rtp_asterisk.c: 0x7f0c40052640 -- Strict RTP switching to RTP target address 178.62.121.41:10972 as source [Aug 18 10:34:23] DEBUG[14830] http.c: match request [ari/channels/212983] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14830] http.c: match request [ari/channels/212983] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14710] channel.c: Channel 0x7f0c780c1d00 'SIP/zvonobot-000000c0' allocated [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14710] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[14830] http.c: match request [ari/channels/212983] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[13861] app.c: One waitfor failed, trying another [Aug 18 10:34:23] DEBUG[14830] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[13227] bridge_native_rtp.c: Bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[13227] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[13227] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[13425] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:23] DEBUG[13425] channel.c: Channel SIP/zvonobot-00000042 setting read format path: ulaw -> alaw [Aug 18 10:34:23] DEBUG[13425] channel.c: Channel SIP/zvonobot-00000042 setting write format path: alaw -> ulaw [Aug 18 10:34:23] DEBUG[13227] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[13227] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[13227] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee is already using the new technology. [Aug 18 10:34:23] DEBUG[13211] stasis.c: Destroying topic. name: cache:70/channel:1629282830.60, detail: [Aug 18 10:34:23] DEBUG[13211] stasis.c: Topic 'cache:70/channel:1629282830.60': 0x7f0c1002d480 destroyed [Aug 18 10:34:23] DEBUG[13211] stasis.c: Destroying topic. name: channel:1629282830.60, detail: [Aug 18 10:34:23] DEBUG[13211] stasis.c: Topic 'channel:1629282830.60': 0x7f0c1000def0 destroyed [Aug 18 10:34:23] WARNING[14657] app.c: No audio available on Recorder/ARI-00000034;1?? [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Finding handler for channels/212983 [Aug 18 10:34:23] VERBOSE[14657] app.c: User hung up [Aug 18 10:34:23] DEBUG[14657] res_stasis_recording.c: 1629282851.317: Recording complete [Aug 18 10:34:23] DEBUG[14657] channel.c: Channel 0x7f0c7804c540 'Recorder/ARI-00000034;1' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Session timer started: 77 - 10f580a044264908688c62534aa40882@159.65.48.104:5060 1768000ms [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14713] channel.c: Channel 0x7f0c8c106b00 'Recorder/ARI-00000046;1' allocated [Aug 18 10:34:23] DEBUG[14831] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14702] channel.c: Channel 0x7f0c3c167210 'SIP/zvonobot-000000bf' allocated [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14702] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[14710] res_stasis.c: calls_0: Subscribing to 213161 [Aug 18 10:34:23] DEBUG[14710] stasis/app.c: Channel '213161' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14710] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[13214] channel.c: Channel 0x7f0c10025b50 'SIP/zvonobot-00000012' destroying [Aug 18 10:34:23] DEBUG[14702] res_stasis.c: calls_0: Subscribing to 213155 [Aug 18 10:34:23] DEBUG[14702] stasis/app.c: Channel '213155' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Outgoing Call for 79821116879 [Aug 18 10:34:23] DEBUG[13227] stasis/control.c: robot_212983, 3f704757-87e2-45e5-8aa9-92ed6ea9feee: Channel was departed from bridge [Aug 18 10:34:23] DEBUG[14713] stasis.c: Creating topic. name: channel:1629282863.506, detail: [Aug 18 10:34:23] DEBUG[14713] stasis.c: Topic 'channel:1629282863.506': 0x7f0c8c03c880 created [Aug 18 10:34:23] DEBUG[14713] stasis.c: Creating topic. name: cache:582/channel:1629282863.506, detail: [Aug 18 10:34:23] DEBUG[14713] stasis.c: Topic 'cache:582/channel:1629282863.506': 0x7f0c8c05cb80 created [Aug 18 10:34:23] DEBUG[13227] stasis/app.c: bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee': is 2 interested in calls_0 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Outgoing Call for 79821116885 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] DEBUG[14831] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:23] DEBUG[14710] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[13221] stasis/control.c: robot_212983: Channel departing bridge [Aug 18 10:34:23] DEBUG[13221] bridge.c: Waiting for 0x7f0c34026cd0(UnicastRTP/127.0.0.1:50397-0x7f0c28011240) bridge thread to die. [Aug 18 10:34:23] DEBUG[14702] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14702] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[13227] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:23] DEBUG[13221] stasis/app.c: channel 'robot_212983': is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Finding handler for 212983 [Aug 18 10:34:23] DEBUG[14831] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '212983': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '212983' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[14831] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[13214] stasis.c: Destroying topic. name: cache:25/channel:212983, detail: [Aug 18 10:34:23] DEBUG[13214] stasis.c: Topic 'cache:25/channel:212983': 0x7f0c10026f10 destroyed [Aug 18 10:34:23] DEBUG[13214] stasis.c: Destroying topic. name: channel:212983, detail: [Aug 18 10:34:23] DEBUG[13214] stasis.c: Topic 'channel:212983': 0x7f0c100281f0 destroyed [Aug 18 10:34:23] DEBUG[13214] channel.c: Channel 0x7f0c2402e210 'Snoop/212983-00000001' destroying [Aug 18 10:34:23] DEBUG[13214] stasis.c: Destroying topic. name: cache:73/channel:1629282830.62, detail: [Aug 18 10:34:23] DEBUG[13214] stasis.c: Topic 'cache:73/channel:1629282830.62': 0x7f0c24007af0 destroyed [Aug 18 10:34:23] DEBUG[14831] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking channels create: Didn't match 212983 [Aug 18 10:34:23] DEBUG[14831] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[13221] channel.c: Channel 0x7f0c2807fb90 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240' hanging up. Refs: 2 [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[14831] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] DEBUG[14831] stasis.c: Creating topic. name: bridge:ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4, detail: [Aug 18 10:34:23] DEBUG[14831] stasis.c: Topic 'bridge:ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4': 0x7f0c74073a90 created [Aug 18 10:34:23] DEBUG[14831] stasis.c: Creating topic. name: cache:583/bridge:ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4, detail: [Aug 18 10:34:23] DEBUG[14831] stasis.c: Topic 'cache:583/bridge:ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4': 0x7f0c74033900 created [Aug 18 10:34:23] DEBUG[14831] bridge_native_rtp.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4' can not use native RTP bridge as two channels are required [Aug 18 10:34:23] DEBUG[14831] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14831] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14831] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:23] DEBUG[14831] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14831] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: calling simple_bridge technology constructor [Aug 18 10:34:23] DEBUG[14831] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: calling simple_bridge technology start [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[13214] stasis.c: Destroying topic. name: channel:1629282830.62, detail: [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:23] DEBUG[13214] stasis.c: Topic 'channel:1629282830.62': 0x7f0c240089d0 destroyed [Aug 18 10:34:23] DEBUG[14831] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14831] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:23] VERBOSE[14833] chan_sip.c: Audio is at 16374 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.507, detail: [Aug 18 10:34:23] DEBUG[14834] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[14834] http.c: HTTP Request URI is /ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/addChannel?channel=213033 [Aug 18 10:34:23] DEBUG[14834] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.507': 0x7f0c3002e830 created [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:584/channel:1629282863.507, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:584/channel:1629282863.507': 0x7f0c300b3540 created [Aug 18 10:34:23] VERBOSE[14832] chan_sip.c: Audio is at 16950 [Aug 18 10:34:23] DEBUG[14830] res_ari.c: Checking channels externalMedia: Didn't match 212983 [Aug 18 10:34:23] DEBUG[14830] res_ari.c: No explicit handler found for 212983. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14834] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14834] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/addChannel] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14834] http.c: Match made with [ari] [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f;received=159.65.48.104 From: ;tag=as42198afd To: ;tag=as3daa4fa6 Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47425851" Content-Length: 0 <-------------> [Aug 18 10:34:23] VERBOSE[14833] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0765581f;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as42198afd [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3daa4fa6 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:584/channel:1629282863.507, detail: [Aug 18 10:34:23] VERBOSE[14832] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:584/channel:1629282863.507': 0x7f0c300b3540 destroyed [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.507, detail: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.507': 0x7f0c3002e830 destroyed [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Finding handler for bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/addChannel [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Finding handler for bridges [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47425851" [Aug 18 10:34:23] VERBOSE[14833] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000012', '', 'Stasis', 'calls_0', 33, 27, 'ANSWERED', 3, '', '212983', '')] [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5993fccb0e95740465a028667804b469@159.65.48.104:5060 (Checking To) --From tag as42198afd --To-tag as3daa4fa6 [Aug 18 10:34:23] VERBOSE[14833] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5993fccb0e95740465a028667804b469@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Finding handler for ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.508, detail: [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Initializing initreq for method INVITE - callid 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:23] DEBUG[14834] res_ari.c: No explicit handler found for ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4. Using wildcard bridgeId. [Aug 18 10:34:23] VERBOSE[14832] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:23] VERBOSE[14832] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (1) INVITE - 5 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Initializing initreq for method INVITE - callid 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (6) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116910@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c3e3a35 Max-Forwards: 70 From: ;tag=as57d38e8b To: Contact: Call-ID: 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1404479865 1404479865 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15334 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (6) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116914@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51f35118 Max-Forwards: 70 From: ;tag=as288a5fb9 To: Contact: Call-ID: 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2135045114 2135045114 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (4) INVITE - 5 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116889@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 Max-Forwards: 70 From: ;tag=as1ac06673 To: Contact: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 722698992 722698992 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116879@178.62.121.41 SIP/2.0 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.508': 0x7f0c3002e830 created [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Finding handler for addChannel [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116885@178.62.121.41 SIP/2.0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:585/channel:1629282863.508, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:585/channel:1629282863.508': 0x7f0c300e03c0 created [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 [Aug 18 10:34:23] DEBUG[14834] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:23] DEBUG[14834] stasis/control.c: 213033: Sending channel add_to_bridge command [Aug 18 10:34:23] DEBUG[13425] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000042 [Aug 18 10:34:23] DEBUG[13425] stasis/control.c: 213033: Adding to bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4 [Aug 18 10:34:23] DEBUG[13425] stasis/app.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 3 [ 52]: From: ;tag=as1cccf2a3 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 3 [ 52]: From: ;tag=as58ae887d [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 6 [ 60]: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:585/channel:1629282863.508, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:585/channel:1629282863.508': 0x7f0c300e03c0 destroyed [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 From: ;tag=as3056f2e0 To: ;tag=as20b331c1 Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" Content-Length: 0 <-------------> [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.508, detail: [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] VERBOSE[14832] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3056f2e0 [Aug 18 10:34:23] DEBUG[14835] bridge_channel.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: 0x7f0c9006b170(SIP/zvonobot-00000042) is joining [Aug 18 10:34:23] DEBUG[14832] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20b331c1 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.508': 0x7f0c3002e830 destroyed [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 6 [ 60]: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 [Aug 18 10:34:23] VERBOSE[14832] dial.c: Called zvonobot/79821116879 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] VERBOSE[14833] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #69 [Aug 18 10:34:23] DEBUG[14833] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <>', '', 's', 'default', 'Snoop/212983-00000001', 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240', 'Stasis', 'calls_0', 29, 29, 'ANSWERED', 3, '', '1629282830.62', '')] [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:23] DEBUG[14835] bridge_channel.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: pushing 0x7f0c9006b170(SIP/zvonobot-00000042) [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:23] VERBOSE[14833] dial.c: Called zvonobot/79821116885 [Aug 18 10:34:23] DEBUG[14408] chan_sip.c: Hangup call SIP/zvonobot-00000095, SIP callid 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14408] res_rtp_asterisk.c: (0x7f0c74055f50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14408] res_rtp_asterisk.c: (0x7f0c74055f50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[14408] channel.c: Channel 0x7f0c7405a610 'SIP/zvonobot-00000095' destroying [Aug 18 10:34:23] DEBUG[14706] channel.c: Channel 0x7f0c400c7ad0 'SIP/zvonobot-000000c1' allocated [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14706] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282863.509, detail: [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:23] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213105': is 0 interested in calls_0 [Aug 18 10:34:23] DEBUG[20620] stasis/app.c: channel '213105' unsubscribed from calls_0 [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.509': 0x7f0c300e03c0 created [Aug 18 10:34:23] DEBUG[14838] http.c: HTTP opening session. Top level [Aug 18 10:34:23] DEBUG[20545] stasis.c: Creating topic. name: cache:586/channel:1629282863.509, detail: [Aug 18 10:34:23] VERBOSE[14835] bridge_channel.c: Channel SIP/zvonobot-00000042 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:586/channel:1629282863.509': 0x7f0c3002e830 created [Aug 18 10:34:23] DEBUG[14708] channel.c: Channel 0x7f0c7003c0c0 'SIP/zvonobot-000000c2' allocated [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:23] DEBUG[14708] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:23] DEBUG[14310] channel.c: Channel 0x7f0c80042760 'Recorder/ARI-0000002e;2' destroying [Aug 18 10:34:23] DEBUG[14310] stasis.c: Destroying topic. name: cache:338/channel:1629282849.292, detail: [Aug 18 10:34:23] DEBUG[14310] stasis.c: Topic 'cache:338/channel:1629282849.292': 0x7f0c800419b0 destroyed [Aug 18 10:34:23] DEBUG[14310] stasis.c: Destroying topic. name: channel:1629282849.292, detail: [Aug 18 10:34:23] DEBUG[14310] stasis.c: Topic 'channel:1629282849.292': 0x7f0c800417d0 destroyed [Aug 18 10:34:23] DEBUG[14838] http.c: HTTP Request URI is /ari/channels/213105 [Aug 18 10:34:23] DEBUG[14357] channel.c: Channel 0x7f0c8c05b2c0 'Snoop/213015-00000014' allocated [Aug 18 10:34:23] DEBUG[14708] res_stasis.c: calls_0: Subscribing to 213162 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" [Aug 18 10:34:23] DEBUG[14112] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83'. Checking compatability for channels 'SIP/zvonobot-00000033' and 'Recorder/ARI-0000002d;2' [Aug 18 10:34:23] DEBUG[14112] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' can not use native RTP bridge as channel 'SIP/zvonobot-00000033' has features which prevent it [Aug 18 10:34:23] DEBUG[14112] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:23] DEBUG[14112] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14112] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:23] DEBUG[14112] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:23] DEBUG[14112] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 is already using the new technology. [Aug 18 10:34:23] DEBUG[14357] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14357] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[14838] http.c: match request [ari/channels/213105] with handler [httpstatus] len 10 [Aug 18 10:34:23] DEBUG[14706] res_stasis.c: calls_0: Subscribing to 213159 [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:23] DEBUG[14838] http.c: match request [ari/channels/213105] with handler [phoneprov] len 9 [Aug 18 10:34:23] DEBUG[14408] stasis.c: Destroying topic. name: cache:359/channel:213105, detail: [Aug 18 10:34:23] DEBUG[14408] stasis.c: Topic 'cache:359/channel:213105': 0x7f0c7405ce10 destroyed [Aug 18 10:34:23] DEBUG[14408] stasis.c: Destroying topic. name: channel:213105, detail: [Aug 18 10:34:23] DEBUG[14408] stasis.c: Topic 'channel:213105': 0x7f0c7405c390 destroyed [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:23] DEBUG[14708] stasis/app.c: Channel '213162' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14838] http.c: match request [ari/channels/213105] with handler [ari] len 3 [Aug 18 10:34:23] DEBUG[14706] stasis/app.c: Channel '213159' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14706] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14708] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:23] DEBUG[14708] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 (Checking To) --From tag as3056f2e0 --To-tag as20b331c1 [Aug 18 10:34:23] DEBUG[14706] http.c: HTTP closing session. Top level [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Stopping retransmission on '6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Outgoing Call for 79821116878 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:23] VERBOSE[14840] chan_sip.c: Audio is at 11726 [Aug 18 10:34:23] VERBOSE[14840] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:23] VERBOSE[14840] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:23] VERBOSE[14840] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Initializing initreq for method INVITE - callid 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116878@178.62.121.41 SIP/2.0 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 3 [ 52]: From: ;tag=as45eb6124 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 6 [ 60]: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:23] VERBOSE[14840] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #67 [Aug 18 10:34:23] DEBUG[14840] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116920@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1 Max-Forwards: 70 From: ;tag=as3056f2e0 To: Contact: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:23] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:23] DEBUG[14838] http.c: Match made with [ari] [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: cache:586/channel:1629282863.509, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'cache:586/channel:1629282863.509': 0x7f0c3002e830 destroyed [Aug 18 10:34:23] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282863.509, detail: [Aug 18 10:34:23] DEBUG[20545] stasis.c: Topic 'channel:1629282863.509': 0x7f0c300e03c0 destroyed [Aug 18 10:34:23] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000095', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213105', '')] [Aug 18 10:34:23] DEBUG[14839] stasis/app.c: Channel '1629282858.408' is 1 interested in calls_0 [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Finding handler for channels/213105 [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Finding handler for channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Finding handler for 213105 [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking channels create: Didn't match 213105 [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:23] DEBUG[14838] res_ari.c: Checking channels externalMedia: Didn't match 213105 [Aug 18 10:34:23] DEBUG[14838] res_ari.c: No explicit handler found for 213105. Using wildcard channelId. [Aug 18 10:34:23] DEBUG[14717] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:23] DEBUG[13857] res_rtp_asterisk.c: (0x7f0c7804a920) DTLS stop [Aug 18 10:34:23] DEBUG[13857] res_rtp_asterisk.c: (0x7f0c7804a920) DTLS srtp - stopped timeout timer' [Aug 18 10:34:23] DEBUG[13857] res_rtp_asterisk.c: (0x7f0c7804a920) ICE RTP transport deallocating [Aug 18 10:34:24] DEBUG[13857] res_rtp_asterisk.c: (0x7f0c7804a920) ICE stopped [Aug 18 10:34:24] DEBUG[13857] rtp_engine.c: Destroyed RTP instance '0x7f0c7804a920' [Aug 18 10:34:24] DEBUG[13857] channel.c: Channel 0x7f0c7807b3b0 'UnicastRTP/127.0.0.1:50430-0x7f0c7804a920' destroying [Aug 18 10:34:23] DEBUG[14843] chan_sip.c: Outgoing Call for 79821116881 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:24] DEBUG[14573] channel.c: Channel 0x7f0c8c034ba0 'Announcer/ARI-0000003d;2' allocated [Aug 18 10:34:24] DEBUG[14573] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:24] DEBUG[14573] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000003d;1' [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:24] VERBOSE[14843] chan_sip.c: Audio is at 11948 [Aug 18 10:34:24] VERBOSE[14843] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:24] VERBOSE[14843] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:24] DEBUG[14717] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:24] VERBOSE[14840] dial.c: Called zvonobot/79821116878 [Aug 18 10:34:24] DEBUG[14844] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) is joining [Aug 18 10:34:24] VERBOSE[14843] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6478ms with no response [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 298 instead [Aug 18 10:34:24] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282864.510, detail: [Aug 18 10:34:24] DEBUG[20545] stasis.c: Topic 'channel:1629282864.510': 0x7f0c3002e830 created [Aug 18 10:34:24] DEBUG[20545] stasis.c: Creating topic. name: cache:587/channel:1629282864.510, detail: [Aug 18 10:34:24] DEBUG[20545] stasis.c: Topic 'cache:587/channel:1629282864.510': 0x7f0c300e03c0 created [Aug 18 10:34:24] DEBUG[20545] stasis.c: Destroying topic. name: cache:587/channel:1629282864.510, detail: [Aug 18 10:34:24] DEBUG[20545] stasis.c: Topic 'cache:587/channel:1629282864.510': 0x7f0c300e03c0 destroyed [Aug 18 10:34:24] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282864.510, detail: [Aug 18 10:34:24] DEBUG[20545] stasis.c: Topic 'channel:1629282864.510': 0x7f0c3002e830 destroyed [Aug 18 10:34:24] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:02', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50430-0x7f0c7804a920', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213023', '')] [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Initializing initreq for method INVITE - callid 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116881@178.62.121.41 SIP/2.0 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 3 [ 52]: From: ;tag=as7eb8ca07 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 6 [ 60]: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:23 GMT [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:24] VERBOSE[14843] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14844] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: pushing 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 [Aug 18 10:34:24] DEBUG[14843] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[13206] res_rtp_asterisk.c: (0x7f0ca402d940) RTP 0x7f0ca402f490 -- Received packet from 178.62.121.41:15212, dropping due to strict RTP protection. [Aug 18 10:34:24] DEBUG[14545] channel.c: Channel 0x7f0cb011fc00 'SIP/zvonobot-000000a4' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a4 - start 1629282857.302437 answer 0.000000 end 1629282864.028435 dur 6.725 bill 1629282864.028 dispo NO ANSWER [Aug 18 10:34:24] DEBUG[14112] audiohook.c: Audiohook 0x7f0c8c051450 has stale audio in its factories. Flushing them both [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6499ms with no response [Aug 18 10:34:24] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] VERBOSE[14843] dial.c: Called zvonobot/79821116881 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:24] DEBUG[14835] bridge_native_rtp.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4' can not use native RTP bridge as two channels are required [Aug 18 10:34:24] DEBUG[14835] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14835] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14835] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14835] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14835] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4 is already using the new technology. [Aug 18 10:34:24] DEBUG[14835] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: 0x7f0c9006b170(SIP/zvonobot-00000042) is joining simple_bridge technology [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:24] DEBUG[20620] stasis/app.c: channel 'robot_213023': is 0 interested in calls_0 [Aug 18 10:34:24] DEBUG[20620] stasis/app.c: channel 'robot_213023' unsubscribed from calls_0 [Aug 18 10:34:24] DEBUG[20620] stasis.c: Destroying topic. name: cache:250/channel:robot_213023, detail: [Aug 18 10:34:24] DEBUG[20620] stasis.c: Topic 'cache:250/channel:robot_213023': 0x7f0c7803b0e0 destroyed [Aug 18 10:34:24] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213023, detail: [Aug 18 10:34:24] DEBUG[20620] stasis.c: Topic 'channel:robot_213023': 0x7f0c780728f0 destroyed [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 From: ;tag=as18b114f0 To: ;tag=as342fd06d Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 392402988 392402988 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14534 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:24] DEBUG[14846] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a5 - start 1629282857.375540 answer 0.000000 end 1629282864.065280 dur 6.689 bill 1629282864.065 dispo NO ANSWER [Aug 18 10:34:24] DEBUG[14846] http.c: HTTP Request URI is /ari/channels/213174?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116866&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14549] channel.c: Channel 0x7f0ca8109f70 'SIP/zvonobot-000000a5' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[14844] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:24] VERBOSE[14844] bridge_channel.c: Channel Announcer/ARI-0000003d;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:24] DEBUG[14835] res_rtp_asterisk.c: (0x7f0c40036750) RTP changing ssrc from 243273542 to 679482383 due to a source change [Aug 18 10:34:24] DEBUG[14844] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846'. Checking compatability for channels 'SIP/zvonobot-00000038' and 'Announcer/ARI-0000003d;2' [Aug 18 10:34:24] DEBUG[14844] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as channel 'SIP/zvonobot-00000038' has features which prevent it [Aug 18 10:34:24] DEBUG[14844] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14844] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14844] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14844] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14844] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:24] DEBUG[14844] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) is joining simple_bridge technology [Aug 18 10:34:24] DEBUG[14844] channel.c: Channel Announcer/ARI-0000003d;2 setting read format path: slin -> slin [Aug 18 10:34:24] DEBUG[14844] channel.c: Channel Announcer/ARI-0000003d;2 setting write format path: slin -> slin [Aug 18 10:34:24] DEBUG[14711] channel.c: Channel 0x7f0c8411eca0 'SIP/zvonobot-000000c3' allocated [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:24] DEBUG[14853] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14834] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:24] DEBUG[14834] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[13425] stasis/app.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4' is 2 interested in calls_0 [Aug 18 10:34:24] DEBUG[14853] http.c: HTTP Request URI is /ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/record?name=213033_cjiiXAUnWZySsSbRJLqGopVrjnYAMqVg&format=wav [Aug 18 10:34:24] DEBUG[14853] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/record] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14853] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/record] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14853] http.c: match request [ari/bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/record] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14853] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:24] DEBUG[14573] res_stasis_playback.c: 1629282858.405: Sending play(sound:silence/2) command [Aug 18 10:34:24] DEBUG[14573] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:24] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14851] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14573] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14146] channel.c: SIP/zvonobot-00000038: Dropping redundant connected line update "" <>. [Aug 18 10:34:24] DEBUG[14851] http.c: HTTP Request URI is /ari/channels/213177?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116863&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14851] http.c: match request [ari/channels/213177] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14851] http.c: match request [ari/channels/213177] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:24] DEBUG[14851] http.c: match request [ari/channels/213177] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:24] DEBUG[14711] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:24] DEBUG[14851] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14711] res_stasis.c: calls_0: Subscribing to 213160 [Aug 18 10:34:24] DEBUG[14711] stasis/app.c: Channel '213160' is 1 interested in calls_0 [Aug 18 10:34:24] DEBUG[14711] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14711] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14846] http.c: match request [ari/channels/213174] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14846] http.c: match request [ari/channels/213174] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14846] http.c: match request [ari/channels/213174] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14846] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14851] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK42c12d33;received=159.65.48.104 [Aug 18 10:34:24] DEBUG[14857] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Finding handler for channels/213177 [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Finding handler for 213177 [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking channels create: Didn't match 213177 [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14851] res_ari.c: Checking channels externalMedia: Didn't match 213177 [Aug 18 10:34:24] DEBUG[14851] res_ari.c: No explicit handler found for 213177. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Finding handler for bridges/ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4/record [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as18b114f0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as342fd06d [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Finding handler for bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Finding handler for ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4 [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14853] res_ari.c: No explicit handler found for ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4. Using wildcard bridgeId. [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Finding handler for record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:24] DEBUG[14853] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:24] DEBUG[14853] stasis.c: Creating topic. name: channel:1629282864.511, detail: [Aug 18 10:34:24] DEBUG[14853] stasis.c: Topic 'channel:1629282864.511': 0x7f0c98029370 created [Aug 18 10:34:24] DEBUG[14853] stasis.c: Creating topic. name: cache:588/channel:1629282864.511, detail: [Aug 18 10:34:24] DEBUG[14853] stasis.c: Topic 'cache:588/channel:1629282864.511': 0x7f0c98095fc0 created [Aug 18 10:34:24] DEBUG[14857] http.c: HTTP Request URI is /ari/channels/213176?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116864&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14857] http.c: match request [ari/channels/213176] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14857] http.c: match request [ari/channels/213176] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14857] http.c: match request [ari/channels/213176] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14857] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14857] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Finding handler for channels/213176 [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Finding handler for 213176 [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking channels create: Didn't match 213176 [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14857] res_ari.c: Checking channels externalMedia: Didn't match 213176 [Aug 18 10:34:24] DEBUG[14857] res_ari.c: No explicit handler found for 213176. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14846] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Finding handler for channels/213174 [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Finding handler for 213174 [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking channels create: Didn't match 213174 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:24] WARNING[13678] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-0000001c;1 [Aug 18 10:34:24] DEBUG[14862] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14856] channel.c: Channel Announcer/ARI-0000003d;1 setting write format path: gsm -> slin [Aug 18 10:34:24] DEBUG[14146] audiohook.c: Audiohook 0x7f0c700a8980 has stale audio in its factories. Flushing them both [Aug 18 10:34:24] DEBUG[14865] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14146] res_rtp_asterisk.c: (0x7f0c38043ba0) RTP ooh, format changed from none to alaw [Aug 18 10:34:24] DEBUG[14865] http.c: HTTP Request URI is /ari/channels/213179?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116861&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14865] http.c: match request [ari/channels/213179] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14146] res_rtp_asterisk.c: (0x7f0c38043ba0) RTCP starting transmission [Aug 18 10:34:24] DEBUG[14865] http.c: match request [ari/channels/213179] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14865] http.c: match request [ari/channels/213179] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14865] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14856] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:24] VERBOSE[14856] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:24] DEBUG[14865] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Finding handler for channels/213179 [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Finding handler for 213179 [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking channels create: Didn't match 213179 [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14865] res_ari.c: Checking channels externalMedia: Didn't match 213179 [Aug 18 10:34:24] DEBUG[14865] res_ari.c: No explicit handler found for 213179. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14862] http.c: HTTP Request URI is /ari/channels/213175?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116865&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Outgoing Call for 79821116880 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:24] VERBOSE[14641] res_rtp_asterisk.c: 0x7f0c180a2570 -- Strict RTP learning complete - Locking on source address 178.62.121.41:17730 [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14846] res_ari.c: Checking channels externalMedia: Didn't match 213174 [Aug 18 10:34:24] DEBUG[14846] res_ari.c: No explicit handler found for 213174. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14862] http.c: match request [ari/channels/213175] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14862] http.c: match request [ari/channels/213175] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14862] http.c: match request [ari/channels/213175] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14862] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14862] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Finding handler for channels/213175 [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Finding handler for 213175 [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking channels create: Didn't match 213175 [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14862] res_ari.c: Checking channels externalMedia: Didn't match 213175 [Aug 18 10:34:24] DEBUG[14862] res_ari.c: No explicit handler found for 213175. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14873] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14873] http.c: HTTP Request URI is /ari/channels/213181?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116859&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14873] http.c: match request [ari/channels/213181] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14873] http.c: match request [ari/channels/213181] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14873] http.c: match request [ari/channels/213181] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14873] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14873] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Finding handler for channels/213181 [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Finding handler for 213181 [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking channels create: Didn't match 213181 [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14873] res_ari.c: Checking channels externalMedia: Didn't match 213181 [Aug 18 10:34:24] DEBUG[14873] res_ari.c: No explicit handler found for 213181. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 392402988 392402988 IN IP4 178.62.121.41 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14534 RTP/AVP 0 8 101 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: = Looking for Call ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 (Checking To) --From tag as18b114f0 --To-tag as342fd06d [Aug 18 10:34:24] DEBUG[14875] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14875] http.c: HTTP Request URI is /ari/channels/213178?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116862&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14875] http.c: match request [ari/channels/213178] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14875] http.c: match request [ari/channels/213178] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14875] http.c: match request [ari/channels/213178] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14875] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14875] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Finding handler for channels/213178 [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14641] res_rtp_asterisk.c: (0x7f0c18094150) RTCP got report of 76 bytes from 178.62.121.41:17731 [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Finding handler for 213178 [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking channels create: Didn't match 213178 [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14875] res_ari.c: Checking channels externalMedia: Didn't match 213178 [Aug 18 10:34:24] DEBUG[14875] res_ari.c: No explicit handler found for 213178. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:24] DEBUG[14874] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14874] http.c: HTTP Request URI is /ari/channels/213180?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116860&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14874] http.c: match request [ari/channels/213180] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14874] http.c: match request [ari/channels/213180] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14876] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14876] http.c: HTTP Request URI is /ari/channels/213182?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116858&callerId=74950493843 [Aug 18 10:34:24] DEBUG[14876] http.c: match request [ari/channels/213182] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[14876] http.c: match request [ari/channels/213182] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14876] http.c: match request [ari/channels/213182] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14876] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14876] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Finding handler for channels/213182 [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:24] DEBUG[14567] channel.c: Channel 0x7f0c7c054de0 'Recorder/ARI-0000003e;2' allocated [Aug 18 10:34:24] DEBUG[14567] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:24] DEBUG[14874] http.c: match request [ari/channels/213180] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Stopping retransmission on '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:24] DEBUG[14874] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14874] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14712] channel.c: Channel 0x7f0c80016c20 'SIP/zvonobot-000000c4' allocated [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:24] DEBUG[14712] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:24] DEBUG[14878] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) is joining [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Strict routing enforced for session 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Finding handler for channels/213180 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Finding handler for channels [Aug 18 10:34:24] VERBOSE[14858] chan_sip.c: Audio is at 16538 [Aug 18 10:34:24] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Finding handler for 213182 [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking channels create: Didn't match 213182 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] VERBOSE[14858] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:24] DEBUG[14879] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: Allocating new SIP dialog for 4be4cb092ae1fe722fc7cc1011215d41@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14851] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c0a2320' [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) RTP allocated port 10056 [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE creating session 0.0.0.0:10056 (10056) [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE create [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE add system candidates [Aug 18 10:34:24] DEBUG[14851] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14851] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE add candidate: 159.65.48.104:10056, 2130706431 [Aug 18 10:34:24] DEBUG[14851] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14851] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE add candidate: 10.131.0.10:10056, 2130706431 [Aug 18 10:34:24] DEBUG[14851] rtp_engine.c: RTP instance '0x7f0c9c0a2320' is setup and ready to go [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) ICE stopped [Aug 18 10:34:24] DEBUG[14851] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14851] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14851] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14851] res_rtp_asterisk.c: (0x7f0c9c0a2320) RTCP setup on RTP instance [Aug 18 10:34:24] VERBOSE[14851] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Finding handler for 213180 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking channels create: Didn't match 213180 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14874] res_ari.c: Checking channels externalMedia: Didn't match 213180 [Aug 18 10:34:24] DEBUG[14874] res_ari.c: No explicit handler found for 213180. Using wildcard channelId. [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14878] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: pushing 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) [Aug 18 10:34:24] DEBUG[14876] res_ari.c: Checking channels externalMedia: Didn't match 213182 [Aug 18 10:34:24] DEBUG[14876] res_ari.c: No explicit handler found for 213182. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117024@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5993aaa8 Max-Forwards: 70 From: ;tag=as18b114f0 To: ;tag=as342fd06d Contact: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:24] DEBUG[14879] http.c: HTTP Request URI is /ari/channels/213183?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116857&callerId=74950493843 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Destroying SIP dialog 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '63e4041b488585c57e57de141ed1835f@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74055f50) DTLS stop [Aug 18 10:34:24] DEBUG[14878] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74055f50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] VERBOSE[14878] bridge_channel.c: Channel Recorder/ARI-0000003e;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:24] VERBOSE[14858] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: Allocating new SIP dialog for 7428e31075190b107420905a54c9b56c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14857] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca4120df0' [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) RTP allocated port 16258 [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE creating session 0.0.0.0:16258 (16258) [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE create [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE add system candidates [Aug 18 10:34:24] DEBUG[14857] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14857] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE add candidate: 159.65.48.104:16258, 2130706431 [Aug 18 10:34:24] DEBUG[14857] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14857] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE add candidate: 10.131.0.10:16258, 2130706431 [Aug 18 10:34:24] DEBUG[14857] rtp_engine.c: RTP instance '0x7f0ca4120df0' is setup and ready to go [Aug 18 10:34:24] DEBUG[14879] http.c: match request [ari/channels/213183] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74055f50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c74055f50) ICE RTP transport deallocating [Aug 18 10:34:24] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c74055f50' [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6607ms with no response [Aug 18 10:34:24] VERBOSE[14858] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14554] channel.c: Channel 0x7f0c88058300 'SIP/zvonobot-000000a7' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[14878] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06'. Checking compatability for channels 'SIP/zvonobot-00000027' and 'Recorder/ARI-0000003e;2' [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a7 - start 1629282857.697349 answer 0.000000 end 1629282864.405990 dur 6.708 bill 1629282864.405 dispo NO ANSWER [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) ICE stopped [Aug 18 10:34:24] DEBUG[14857] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14857] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14857] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14857] res_rtp_asterisk.c: (0x7f0ca4120df0) RTCP setup on RTP instance [Aug 18 10:34:24] VERBOSE[14857] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14857] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14857] chan_sip.c: SIP call-id changed from '7428e31075190b107420905a54c9b56c@127.0.1.1:5060' to '45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14857] stasis.c: Creating topic. name: channel:213176, detail: [Aug 18 10:34:24] DEBUG[14857] stasis.c: Topic 'channel:213176': 0x7f0ca40f5de0 created [Aug 18 10:34:24] DEBUG[14857] stasis.c: Creating topic. name: cache:589/channel:213176, detail: [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:24] DEBUG[14878] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' can not use native RTP bridge as could not get details [Aug 18 10:34:24] DEBUG[14879] http.c: match request [ari/channels/213183] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Initializing initreq for method INVITE - callid 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116880@178.62.121.41 SIP/2.0 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 3 [ 52]: From: ;tag=as71bf7e35 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 6 [ 60]: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:24 GMT [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:24] VERBOSE[14858] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Aug 18 10:34:24] DEBUG[14858] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:24] DEBUG[14851] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6602ms with no response [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14555] channel.c: Channel 0x2c522e0 'SIP/zvonobot-000000a8' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a8 - start 1629282857.728848 answer 0.000000 end 1629282864.439160 dur 6.710 bill 1629282864.439 dispo NO ANSWER [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (5) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116887@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 Max-Forwards: 70 From: ;tag=as7b595413 To: Contact: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 184815596 184815596 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10240 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (3) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6441ms with no response [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14712] res_stasis.c: calls_0: Subscribing to 213163 [Aug 18 10:34:24] DEBUG[14712] stasis/app.c: Channel '213163' is 1 interested in calls_0 [Aug 18 10:34:24] DEBUG[14879] http.c: match request [ari/channels/213183] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64;received=159.65.48.104 From: ;tag=as293a990c To: ;tag=as401cd277 Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48bad81c" Content-Length: 0 <-------------> [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK479fdc64;received=159.65.48.104 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as293a990c [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as401cd277 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48bad81c" [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 (Checking To) --From tag as293a990c --To-tag as401cd277 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63e831cc0986e7ef00ad4da2513324d2@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (3) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[14878] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14878] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14878] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14878] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14878] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06 is already using the new technology. [Aug 18 10:34:24] DEBUG[14878] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) is joining simple_bridge technology [Aug 18 10:34:24] DEBUG[14878] channel.c: Channel Recorder/ARI-0000003e;2 setting read format path: slin -> slin [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:24] DEBUG[14712] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14857] stasis.c: Topic 'cache:589/channel:213176': 0x7f0ca407d3c0 created [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14879] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] DEBUG[14712] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: Allocating new SIP dialog for 68cecbbe09b82bce52c0db166d691b03@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14146] res_rtp_asterisk.c: (0x7f0c38043ba0) RTP audio difference is 664, ms is 103 [Aug 18 10:34:24] DEBUG[13762] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14879] http.c: HTTP consuming request body [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Outgoing Call for 79821116877 [Aug 18 10:34:24] DEBUG[14579] channel.c: Channel 0x7f0c940b0650 'Announcer/ARI-0000003f;2' allocated [Aug 18 10:34:24] DEBUG[14579] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:24] DEBUG[14579] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000003f;1' [Aug 18 10:34:24] DEBUG[14882] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c940b36c0(Announcer/ARI-0000003f;2) is joining [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] VERBOSE[14858] dial.c: Called zvonobot/79821116880 [Aug 18 10:34:24] DEBUG[14723] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:24] DEBUG[14723] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14878] channel.c: Channel SIP/zvonobot-00000027 setting write format path: slin -> ulaw [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Destroying SIP dialog 63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[14865] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb4030170' [Aug 18 10:34:24] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] DEBUG[14878] channel.c: Channel SIP/zvonobot-00000027 setting read format path: ulaw -> slin [Aug 18 10:34:24] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Finding handler for channels/213183 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:24] VERBOSE[14880] chan_sip.c: Audio is at 16822 [Aug 18 10:34:24] VERBOSE[14880] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '63c2fb4202377d7867d5709409bbc45f@159.65.48.104:5060' Method: BYE [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: Allocating new SIP dialog for 666a5c556dbf87271ce866b9618e5a4b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14873] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c77e90' [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) RTP allocated port 19846 [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE creating session 0.0.0.0:19846 (19846) [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE create [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE add system candidates [Aug 18 10:34:24] DEBUG[14873] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14873] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:24] DEBUG[14851] chan_sip.c: SIP call-id changed from '4be4cb092ae1fe722fc7cc1011215d41@127.0.1.1:5060' to '63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) RTP allocated port 18760 [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE creating session 0.0.0.0:18760 (18760) [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE create [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: Allocating new SIP dialog for 3ce950a64f53e307610e03816b4d1cad@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14878] channel.c: Channel Recorder/ARI-0000003e;2 setting write format path: slin -> slin [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS stop [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1007bd40) ICE RTP transport deallocating [Aug 18 10:34:24] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c1007bd40' [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6430ms with no response [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (5) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1822830223 1822830223 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14846] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca8109680' [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE add candidate: 159.65.48.104:19846, 2130706431 [Aug 18 10:34:24] DEBUG[14873] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14873] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE add system candidates [Aug 18 10:34:24] DEBUG[14865] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14865] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE add candidate: 159.65.48.104:18760, 2130706431 [Aug 18 10:34:24] DEBUG[14865] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14865] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE add candidate: 10.131.0.10:18760, 2130706431 [Aug 18 10:34:24] DEBUG[14851] stasis.c: Creating topic. name: channel:213177, detail: [Aug 18 10:34:24] DEBUG[14851] stasis.c: Topic 'channel:213177': 0x7f0c9c023fd0 created [Aug 18 10:34:24] DEBUG[14851] stasis.c: Creating topic. name: cache:590/channel:213177, detail: [Aug 18 10:34:24] DEBUG[14851] stasis.c: Topic 'cache:590/channel:213177': 0x7f0c9c046c10 created [Aug 18 10:34:24] DEBUG[14865] rtp_engine.c: RTP instance '0x7f0cb4030170' is setup and ready to go [Aug 18 10:34:24] DEBUG[14585] channel.c: Channel 0x7f0c2413d0a0 'SIP/zvonobot-00000096' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000096 - start 1629282857.956147 answer 0.000000 end 1629282864.593334 dur 6.637 bill 1629282864.593 dispo NO ANSWER [Aug 18 10:34:24] VERBOSE[14880] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Finding handler for 213183 [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking channels create: Didn't match 213183 [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14879] res_ari.c: Checking channels externalMedia: Didn't match 213183 [Aug 18 10:34:24] DEBUG[14879] res_ari.c: No explicit handler found for 213183. Using wildcard channelId. [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 From: ;tag=as601f237f To: ;tag=as1ebdd48c Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54115709;received=159.65.48.104 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as601f237f [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1ebdd48c [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE add candidate: 10.131.0.10:19846, 2130706431 [Aug 18 10:34:24] DEBUG[14873] rtp_engine.c: RTP instance '0x2c77e90' is setup and ready to go [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) ICE stopped [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) RTP allocated port 12308 [Aug 18 10:34:24] DEBUG[14873] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14873] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 (Checking To) --From tag as601f237f --To-tag as1ebdd48c [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4fc2734566c8b6a6076d4d242c7d32bd@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 486 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (4) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:24] VERBOSE[14880] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE creating session 0.0.0.0:12308 (12308) [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE create [Aug 18 10:34:24] DEBUG[14873] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE add system candidates [Aug 18 10:34:24] DEBUG[14846] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) ICE stopped [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (5) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:24] DEBUG[14846] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE add candidate: 159.65.48.104:12308, 2130706431 [Aug 18 10:34:24] DEBUG[14846] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14846] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE add candidate: 10.131.0.10:12308, 2130706431 [Aug 18 10:34:24] DEBUG[14846] rtp_engine.c: RTP instance '0x7f0ca8109680' is setup and ready to go [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) ICE stopped [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 493433322 493433322 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12660 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14865] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14567] res_stasis_recording.c: 1629282858.407: Sending record(213002_SToxhQcamuWvHrRajlYeasVIVVuLUkoN.wav) command [Aug 18 10:34:24] DEBUG[14567] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:24] DEBUG[14567] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14873] res_rtp_asterisk.c: (0x2c77e90) RTCP setup on RTP instance [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: Allocating new SIP dialog for 496e0a167d45cb67712c7e121a6e7240@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14875] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c08081750' [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) RTP allocated port 18924 [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE creating session 0.0.0.0:18924 (18924) [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE create [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE add system candidates [Aug 18 10:34:24] DEBUG[14875] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14875] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE add candidate: 159.65.48.104:18924, 2130706431 [Aug 18 10:34:24] DEBUG[14875] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14875] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE add candidate: 10.131.0.10:18924, 2130706431 [Aug 18 10:34:24] DEBUG[14875] rtp_engine.c: RTP instance '0x7f0c08081750' is setup and ready to go [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) ICE stopped [Aug 18 10:34:24] DEBUG[14875] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14875] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14875] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] VERBOSE[14873] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14873] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14865] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14865] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14865] res_rtp_asterisk.c: (0x7f0cb4030170) RTCP setup on RTP instance [Aug 18 10:34:24] VERBOSE[14865] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14865] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14865] chan_sip.c: SIP call-id changed from '68cecbbe09b82bce52c0db166d691b03@127.0.1.1:5060' to '6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14882] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: pushing 0x7f0c940b36c0(Announcer/ARI-0000003f;2) [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14873] chan_sip.c: SIP call-id changed from '666a5c556dbf87271ce866b9618e5a4b@127.0.1.1:5060' to '447a50ba7810af590f60b91a27726642@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Initializing initreq for method INVITE - callid 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116877@178.62.121.41 SIP/2.0 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 3 [ 52]: From: ;tag=as06a66f45 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 6 [ 60]: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:24 GMT [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:24] VERBOSE[14880] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #100 [Aug 18 10:34:24] DEBUG[14880] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Destroying SIP dialog 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14885] http.c: HTTP opening session. Top level [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:24] DEBUG[14846] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14846] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14846] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14846] res_rtp_asterisk.c: (0x7f0ca8109680) RTCP setup on RTP instance [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS stop [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: Allocating new SIP dialog for 6e3330d17f5d8a2b2456e455747ddc84@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14874] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1003f1f0' [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) RTP allocated port 13594 [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE creating session 0.0.0.0:13594 (13594) [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE create [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE add system candidates [Aug 18 10:34:24] DEBUG[14874] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14874] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE add candidate: 159.65.48.104:13594, 2130706431 [Aug 18 10:34:24] DEBUG[14874] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14885] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:24] DEBUG[14865] stasis.c: Creating topic. name: channel:213179, detail: [Aug 18 10:34:24] DEBUG[14865] stasis.c: Topic 'channel:213179': 0x7f0cb405a0e0 created [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:24] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0001f70) ICE RTP transport deallocating [Aug 18 10:34:24] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb0001f70' [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14865] stasis.c: Creating topic. name: cache:591/channel:213179, detail: [Aug 18 10:34:24] DEBUG[14865] stasis.c: Topic 'cache:591/channel:213179': 0x7f0cb40428f0 created [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14885] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:24] VERBOSE[14846] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14875] res_rtp_asterisk.c: (0x7f0c08081750) RTCP setup on RTP instance [Aug 18 10:34:24] DEBUG[14856] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14874] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14665] app.c: One waitfor failed, trying another [Aug 18 10:34:24] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14846] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14846] chan_sip.c: SIP call-id changed from '3ce950a64f53e307610e03816b4d1cad@127.0.1.1:5060' to '1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14846] stasis.c: Creating topic. name: channel:213174, detail: [Aug 18 10:34:24] DEBUG[14846] stasis.c: Topic 'channel:213174': 0x7f0ca8034d80 created [Aug 18 10:34:24] DEBUG[14846] stasis.c: Creating topic. name: cache:592/channel:213174, detail: [Aug 18 10:34:24] DEBUG[14846] stasis.c: Topic 'cache:592/channel:213174': 0x7f0ca80e3440 created [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:24] DEBUG[14873] stasis.c: Creating topic. name: channel:213181, detail: [Aug 18 10:34:24] DEBUG[14873] stasis.c: Topic 'channel:213181': 0x2c84720 created [Aug 18 10:34:24] DEBUG[14873] stasis.c: Creating topic. name: cache:593/channel:213181, detail: [Aug 18 10:34:24] DEBUG[14873] stasis.c: Topic 'cache:593/channel:213181': 0x2c23e50 created [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (5) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2000377496 2000377496 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15752 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] VERBOSE[14880] dial.c: Called zvonobot/79821116877 [Aug 18 10:34:24] DEBUG[14885] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE add candidate: 10.131.0.10:13594, 2130706431 [Aug 18 10:34:24] DEBUG[14874] rtp_engine.c: RTP instance '0x7f0c1003f1f0' is setup and ready to go [Aug 18 10:34:24] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14885] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14885] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[14728] stasis.c: Creating topic. name: channel:1629282864.517, detail: [Aug 18 10:34:24] DEBUG[14728] stasis.c: Topic 'channel:1629282864.517': 0x7f0c08081d10 created [Aug 18 10:34:24] DEBUG[14728] stasis.c: Creating topic. name: cache:594/channel:1629282864.517, detail: [Aug 18 10:34:24] DEBUG[14728] stasis.c: Topic 'cache:594/channel:1629282864.517': 0x7f0c08081ec0 created [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) ICE stopped [Aug 18 10:34:24] DEBUG[14874] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14874] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14874] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14874] res_rtp_asterisk.c: (0x7f0c1003f1f0) RTCP setup on RTP instance [Aug 18 10:34:24] VERBOSE[14874] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14874] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] VERBOSE[14875] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14875] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14882] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:24] VERBOSE[14882] bridge_channel.c: Channel Announcer/ARI-0000003f;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:24] DEBUG[14884] app.c: play_and_record: , /var/spool/asterisk/recording/213002_SToxhQcamuWvHrRajlYeasVIVVuLUkoN, 'wav' [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: Allocating new SIP dialog for 36f791802ec095f9779e8ac646b656ea@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:24] DEBUG[14884] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14874] chan_sip.c: SIP call-id changed from '6e3330d17f5d8a2b2456e455747ddc84@127.0.1.1:5060' to '2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14876] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c1c1490d0' [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) RTP allocated port 17486 [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE creating session 0.0.0.0:17486 (17486) [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE create [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96;received=159.65.48.104 From: ;tag=as14ba6e32 To: ;tag=as60c8efa2 Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7aa13769" Content-Length: 0 <-------------> [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14882] bridge.c: Chose bridge technology softmix [Aug 18 10:34:24] VERBOSE[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: switching from simple_bridge technology to softmix [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling softmix technology constructor [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: moving 0x2c6fb50(SIP/zvonobot-00000001) to dummy bridge temporarily [Aug 18 10:34:24] DEBUG[14606] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:24] DEBUG[14444] channel.c: Channel 0x7f0c20015500 'Announcer/ARI-00000031;2' destroying [Aug 18 10:34:24] DEBUG[14875] chan_sip.c: SIP call-id changed from '496e0a167d45cb67712c7e121a6e7240@127.0.1.1:5060' to '0b26716f0c119a15755932c124bb341d@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14875] stasis.c: Creating topic. name: channel:213178, detail: [Aug 18 10:34:24] DEBUG[14875] stasis.c: Topic 'channel:213178': 0x7f0c08075230 created [Aug 18 10:34:24] DEBUG[14875] stasis.c: Creating topic. name: cache:595/channel:213178, detail: [Aug 18 10:34:24] DEBUG[14875] stasis.c: Topic 'cache:595/channel:213178': 0x7f0c0804e450 created [Aug 18 10:34:24] DEBUG[14602] channel.c: Channel 0x7f0c1003bdc0 'Recorder/ARI-00000040;2' allocated [Aug 18 10:34:24] VERBOSE[14884] app.c: x=0, open writing: /var/spool/asterisk/recording/213002_SToxhQcamuWvHrRajlYeasVIVVuLUkoN format: wav, 0x7f0c280e6450 [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: moving 0x7f0c3c10a240(Recorder/ARI-0000002c;2) to dummy bridge temporarily [Aug 18 10:34:24] DEBUG[14606] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is leaving simple_bridge technology (dummy) [Aug 18 10:34:24] DEBUG[14602] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:24] DEBUG[14888] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c100718c0(Recorder/ARI-00000040;2) is joining [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96;received=159.65.48.104 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14ba6e32 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as60c8efa2 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Finding handler for bridges [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:24] DEBUG[14531] channel.c: Channel 0x7f0c40071ab0 'Recorder/ARI-00000033;1' destroying [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:24] DEBUG[14522] bridge_channel.c: Setting 0x7f0c4007ef90(Recorder/ARI-00000033;2) state from:0 to:1 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7aa13769" [Aug 18 10:34:24] DEBUG[14444] stasis.c: Destroying topic. name: cache:367/channel:1629282851.318, detail: [Aug 18 10:34:24] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14444] stasis.c: Topic 'cache:367/channel:1629282851.318': 0x7f0c20061220 destroyed [Aug 18 10:34:24] DEBUG[14444] stasis.c: Destroying topic. name: channel:1629282851.318, detail: [Aug 18 10:34:24] DEBUG[14444] stasis.c: Topic 'channel:1629282851.318': 0x7f0c20076450 destroyed [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE add system candidates [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:24] DEBUG[14876] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:24] DEBUG[14876] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE add candidate: 159.65.48.104:17486, 2130706431 [Aug 18 10:34:24] DEBUG[14876] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:24] DEBUG[14876] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE add candidate: 10.131.0.10:17486, 2130706431 [Aug 18 10:34:24] DEBUG[14876] rtp_engine.c: RTP instance '0x7f0c1c1490d0' is setup and ready to go [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology stop [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c940b36c0(Announcer/ARI-0000003f;2) is joining softmix technology [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Announcer/ARI-0000003f;2: [Aug 18 10:34:24] DEBUG[14882] channel.c: Channel Announcer/ARI-0000003f;2 setting write format path: slin -> slin [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Announcer/ARI-0000003f;2: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Announcer/ARI-0000003f;2: Not in SFU mode [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is joining softmix technology [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: SIP/zvonobot-00000001: [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: SIP/zvonobot-00000001: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: SIP/zvonobot-00000001: Not in SFU mode [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is joining softmix technology [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Recorder/ARI-0000002c;2: [Aug 18 10:34:24] DEBUG[14882] channel.c: Channel Recorder/ARI-0000002c;2 setting write format path: slin -> slin [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:24] DEBUG[14882] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Recorder/ARI-0000002c;2: [Aug 18 10:34:24] DEBUG[14882] bridge_softmix.c: Recorder/ARI-0000002c;2: Not in SFU mode [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling softmix technology start [Aug 18 10:34:24] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology destructor [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) ICE stopped [Aug 18 10:34:24] DEBUG[14876] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:24] DEBUG[14876] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:24] DEBUG[14876] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Finding handler for bridges [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] DEBUG[14522] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: pulling 0x7f0c4007ef90(Recorder/ARI-00000033;2) [Aug 18 10:34:24] VERBOSE[14522] bridge_channel.c: Channel Recorder/ARI-00000033;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:24] DEBUG[14522] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c4007ef90(Recorder/ARI-00000033;2) is leaving simple_bridge technology [Aug 18 10:34:24] DEBUG[14874] stasis.c: Creating topic. name: channel:213180, detail: [Aug 18 10:34:24] DEBUG[14531] stasis.c: Destroying topic. name: cache:364/channel:1629282851.315, detail: [Aug 18 10:34:24] DEBUG[14531] stasis.c: Topic 'cache:364/channel:1629282851.315': 0x7f0c4005a4c0 destroyed [Aug 18 10:34:24] DEBUG[14531] stasis.c: Destroying topic. name: channel:1629282851.315, detail: [Aug 18 10:34:24] DEBUG[14531] stasis.c: Topic 'channel:1629282851.315': 0x7f0c400a40e0 destroyed [Aug 18 10:34:24] DEBUG[14874] stasis.c: Topic 'channel:213180': 0x7f0c10045520 created [Aug 18 10:34:24] DEBUG[14874] stasis.c: Creating topic. name: cache:596/channel:213180, detail: [Aug 18 10:34:24] DEBUG[14522] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as two channels are required [Aug 18 10:34:24] DEBUG[14522] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14522] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:24] DEBUG[14522] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14876] res_rtp_asterisk.c: (0x7f0c1c1490d0) RTCP setup on RTP instance [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: = Looking for Call ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 (Checking To) --From tag as14ba6e32 --To-tag as60c8efa2 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Stopping retransmission on '279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116930@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96 Max-Forwards: 70 From: ;tag=as14ba6e32 To: Contact: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] WARNING[14724] app.c: No audio available on Recorder/ARI-00000039;1?? [Aug 18 10:34:24] VERBOSE[14724] app.c: User hung up [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:24] DEBUG[14522] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14724] res_stasis_recording.c: 1629282854.365: Recording complete [Aug 18 10:34:24] DEBUG[14724] channel.c: Channel 0x7f0c2008f410 'Recorder/ARI-00000039;1' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[14522] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d is already using the new technology. [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:24] DEBUG[14885] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:24] DEBUG[14885] stasis.c: Creating topic. name: bridge:68b325fe-a67d-4a26-bbd5-dea36accaa73, detail: [Aug 18 10:34:24] DEBUG[14885] stasis.c: Topic 'bridge:68b325fe-a67d-4a26-bbd5-dea36accaa73': 0x7f0c3403b2b0 created [Aug 18 10:34:24] DEBUG[14885] stasis.c: Creating topic. name: cache:597/bridge:68b325fe-a67d-4a26-bbd5-dea36accaa73, detail: [Aug 18 10:34:24] DEBUG[14885] stasis.c: Topic 'cache:597/bridge:68b325fe-a67d-4a26-bbd5-dea36accaa73': 0x7f0c3403b360 created [Aug 18 10:34:24] DEBUG[14885] bridge_native_rtp.c: Bridge '68b325fe-a67d-4a26-bbd5-dea36accaa73' can not use native RTP bridge as two channels are required [Aug 18 10:34:24] DEBUG[14885] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14885] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14885] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:24] DEBUG[14885] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14885] bridge.c: Bridge 68b325fe-a67d-4a26-bbd5-dea36accaa73: calling simple_bridge technology constructor [Aug 18 10:34:24] DEBUG[14885] bridge.c: Bridge 68b325fe-a67d-4a26-bbd5-dea36accaa73: calling simple_bridge technology start [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6405ms with no response [Aug 18 10:34:24] DEBUG[14888] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: pushing 0x7f0c100718c0(Recorder/ARI-00000040;2) [Aug 18 10:34:24] WARNING[20585] chan_sip.c: Hanging up call 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14522] channel.c: Channel 0x7f0c400b0ff0 'Recorder/ARI-00000033;2' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[14874] stasis.c: Topic 'cache:596/channel:213180': 0x7f0c10001810 created [Aug 18 10:34:24] DEBUG[14885] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:24] DEBUG[14889] http.c: HTTP opening session. Top level [Aug 18 10:34:24] DEBUG[13638] app.c: One waitfor failed, trying another [Aug 18 10:34:24] DEBUG[14885] http.c: HTTP closing session. Top level [Aug 18 10:34:24] VERBOSE[14876] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:24] DEBUG[14876] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:24] DEBUG[14876] chan_sip.c: SIP call-id changed from '36f791802ec095f9779e8ac646b656ea@127.0.1.1:5060' to '21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060' [Aug 18 10:34:24] DEBUG[14610] channel.c: Channel 0x7f0cb4063e00 'SIP/zvonobot-000000a9' hanging up. Refs: 2 [Aug 18 10:34:24] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a9 - start 1629282858.277536 answer 0.000000 end 1629282864.822496 dur 6.544 bill 1629282864.822 dispo NO ANSWER [Aug 18 10:34:24] DEBUG[14889] http.c: HTTP Request URI is /ari/channels/213002/snoop?app=calls_0&spy=in [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:24] DEBUG[14889] http.c: match request [ari/channels/213002/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14889] http.c: match request [ari/channels/213002/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:24] DEBUG[14889] http.c: match request [ari/channels/213002/snoop] with handler [ari] len 3 [Aug 18 10:34:24] DEBUG[14876] stasis.c: Creating topic. name: channel:213182, detail: [Aug 18 10:34:24] DEBUG[14889] http.c: Match made with [ari] [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (2) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:24] DEBUG[14692] channel.c: Channel 0x7f0c2c00d570 'SIP/zvonobot-000000c5' allocated [Aug 18 10:34:24] DEBUG[14888] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:24] VERBOSE[14888] bridge_channel.c: Channel Recorder/ARI-00000040;2 joined 'simple_bridge' stasis-bridge <79f92216-f8f4-49dd-85f1-f154853e1fd1> [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14685] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTCP got report of 76 bytes from 178.62.121.41:15869 [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:24] DEBUG[14692] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:24] VERBOSE[14685] res_rtp_asterisk.c: 0x7f0ca4101500 -- Strict RTP learning complete - Locking on source address 178.62.121.41:15868 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (5) INVITE - 5 [Aug 18 10:34:24] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Finding handler for channels/213002/snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Finding handler for channels [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:24] DEBUG[14876] stasis.c: Topic 'channel:213182': 0x7f0c1c073600 created [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1216361947 1216361947 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10316 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Finding handler for 213002 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channels create: Didn't match 213002 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (1) INVITE - 5 [Aug 18 10:34:24] DEBUG[14888] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1'. Checking compatability for channels 'SIP/zvonobot-00000002' and 'Recorder/ARI-00000040;2' [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channels externalMedia: Didn't match 213002 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:24] DEBUG[14889] res_ari.c: No explicit handler found for 213002. Using wildcard channelId. [Aug 18 10:34:24] DEBUG[14876] stasis.c: Creating topic. name: cache:598/channel:213182, detail: [Aug 18 10:34:24] DEBUG[14579] res_stasis_playback.c: 1629282858.410: Sending play(sound:silence/2) command [Aug 18 10:34:24] DEBUG[14888] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' can not use native RTP bridge as could not get details [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Finding handler for snoop [Aug 18 10:34:24] DEBUG[14579] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:24] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:24] DEBUG[14579] http.c: HTTP closing session. Top level [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14888] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:24] DEBUG[14876] stasis.c: Topic 'cache:598/channel:213182': 0x7f0c1c06bfe0 created [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:24] DEBUG[14888] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:24] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:24] DEBUG[14888] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:24] DEBUG[14889] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:24] DEBUG[14887] bridge_softmix.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: starting mixing thread [Aug 18 10:34:24] DEBUG[14888] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:24] DEBUG[14888] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1 is already using the new technology. [Aug 18 10:34:24] DEBUG[14888] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c100718c0(Recorder/ARI-00000040;2) is joining simple_bridge technology [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (3) INVITE - 5 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:24] DEBUG[14890] channel.c: Channel Announcer/ARI-0000003f;1 setting write format path: gsm -> slin [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14697] channel.c: Channel 0x7f0c30130ce0 'SIP/zvonobot-000000c7' allocated [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:24] DEBUG[14697] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:24] DEBUG[14095] audiohook.c: Audiohook 0x7f0c3010f100 has stale audio in its factories. Flushing them both [Aug 18 10:34:24] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTP ooh, format changed from none to ulaw [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (3) INVITE - 5 [Aug 18 10:34:24] DEBUG[14888] channel.c: Channel Recorder/ARI-00000040;2 setting read format path: slin -> slin [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[14890] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:24] DEBUG[14888] channel.c: Channel SIP/zvonobot-00000002 setting write format path: slin -> alaw [Aug 18 10:34:24] VERBOSE[14890] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (3) INVITE - 5 [Aug 18 10:34:24] DEBUG[14888] channel.c: Channel SIP/zvonobot-00000002 setting read format path: alaw -> slin [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:24] DEBUG[14888] channel.c: Channel Recorder/ARI-00000040;2 setting write format path: slin -> slin [Aug 18 10:34:24] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:24] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:24] DEBUG[14697] res_stasis.c: calls_0: Subscribing to 213157 [Aug 18 10:34:24] DEBUG[14697] stasis/app.c: Channel '213157' is 1 interested in calls_0 [Aug 18 10:34:24] DEBUG[14697] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:24] DEBUG[14697] http.c: HTTP closing session. Top level [Aug 18 10:34:24] DEBUG[14891] chan_sip.c: Outgoing Call for 79821116883 [Aug 18 10:34:24] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:24] DEBUG[14692] res_stasis.c: calls_0: Subscribing to 213156 [Aug 18 10:34:25] DEBUG[14692] stasis/app.c: Channel '213156' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Outgoing Call for 79821116884 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[14705] channel.c: Channel 0x7f0c3803b8c0 'SIP/zvonobot-000000c6' allocated [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14705] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[14692] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[14468] chan_sip.c: Hangup call SIP/zvonobot-00000098, SIP callid 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14468] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:25] DEBUG[14468] res_rtp_asterisk.c: (0x7f0c340b9d00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14468] res_rtp_asterisk.c: (0x7f0c340b9d00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14468] channel.c: Channel 0x7f0c340f0030 'SIP/zvonobot-00000098' destroying [Aug 18 10:34:25] DEBUG[14466] chan_sip.c: Hangup call SIP/zvonobot-00000097, SIP callid 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14471] chan_sip.c: Hangup call SIP/zvonobot-0000009a, SIP callid 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14471] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:25] DEBUG[14475] chan_sip.c: Hangup call SIP/zvonobot-00000099, SIP callid 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14692] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14466] res_rtp_asterisk.c: (0x7f0c280cff50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14466] res_rtp_asterisk.c: (0x7f0c280cff50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[14471] res_rtp_asterisk.c: (0x7f0c7c0c2520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[13423] chan_sip.c: Hangup call SIP/zvonobot-00000043, SIP callid 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14761] channel.c: Channel 0x7f0ca4043bc0 'Recorder/ARI-00000047;1' allocated [Aug 18 10:34:25] DEBUG[14471] res_rtp_asterisk.c: (0x7f0c7c0c2520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14471] channel.c: Channel 0x7f0c7c0b37d0 'SIP/zvonobot-0000009a' destroying [Aug 18 10:34:25] DEBUG[14475] res_rtp_asterisk.c: (0x7f0c3c11fbd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14475] res_rtp_asterisk.c: (0x7f0c3c11fbd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14475] channel.c: Channel 0x7f0c3c1318b0 'SIP/zvonobot-00000099' destroying [Aug 18 10:34:25] DEBUG[14466] channel.c: Channel 0x7f0c280d8530 'SIP/zvonobot-00000097' destroying [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213117': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213117' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[13423] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14761] stasis.c: Creating topic. name: channel:1629282865.521, detail: [Aug 18 10:34:25] DEBUG[13192] res_rtp_asterisk.c: (0x7f0c8c020490) RTP 0x7f0c8c0246a0 -- Received packet from 178.62.121.41:17184, dropping due to strict RTP protection. [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: Allocating new SIP dialog for 236f5bae10533ca0146105b1586ae542@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:25] DEBUG[14862] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac096810' [Aug 18 10:34:25] DEBUG[13423] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:25] DEBUG[14761] stasis.c: Topic 'channel:1629282865.521': 0x7f0ca4042250 created [Aug 18 10:34:25] DEBUG[14761] stasis.c: Creating topic. name: cache:599/channel:1629282865.521, detail: [Aug 18 10:34:25] DEBUG[14761] stasis.c: Topic 'cache:599/channel:1629282865.521': 0x7f0ca4033f40 created [Aug 18 10:34:25] DEBUG[14475] stasis.c: Destroying topic. name: cache:383/channel:213117, detail: [Aug 18 10:34:25] DEBUG[14475] stasis.c: Topic 'cache:383/channel:213117': 0x7f0c3c133fd0 destroyed [Aug 18 10:34:25] DEBUG[14475] stasis.c: Destroying topic. name: channel:213117, detail: [Aug 18 10:34:25] DEBUG[14475] stasis.c: Topic 'channel:213117': 0x7f0c3c118690 destroyed [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14614] channel.c: Channel 0x7f0c30037010 'Recorder/ARI-00000041;1' allocated [Aug 18 10:34:25] DEBUG[14887] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[13423] channel.c: Channel 0x7f0c280bd370 'SIP/zvonobot-00000043' destroying [Aug 18 10:34:25] DEBUG[14893] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213118': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213118' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[14894] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14894] http.c: HTTP Request URI is /ari/channels/213118 [Aug 18 10:34:25] DEBUG[14894] http.c: match request [ari/channels/213118] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14894] http.c: match request [ari/channels/213118] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14894] http.c: match request [ari/channels/213118] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: Allocating new SIP dialog for 451071ca38cc328204fe5d2b2ad4fe1e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:25] DEBUG[14705] res_stasis.c: calls_0: Subscribing to 213158 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.523, detail: [Aug 18 10:34:25] DEBUG[14893] http.c: HTTP Request URI is /ari/channels/213117 [Aug 18 10:34:25] DEBUG[14614] stasis.c: Creating topic. name: channel:1629282865.522, detail: [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] DEBUG[14894] http.c: Match made with [ari] [Aug 18 10:34:25] VERBOSE[14891] chan_sip.c: Audio is at 16210 [Aug 18 10:34:25] VERBOSE[14891] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] VERBOSE[14891] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] VERBOSE[14891] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) RTP allocated port 16814 [Aug 18 10:34:25] DEBUG[14614] stasis.c: Topic 'channel:1629282865.522': 0x7f0c300b32b0 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.523': 0x7f0c300fba90 created [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:384/channel:213118, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:384/channel:213118': 0x7f0c7c0a3370 destroyed [Aug 18 10:34:25] DEBUG[14705] stasis/app.c: Channel '213158' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE creating session 0.0.0.0:16814 (16814) [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE create [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE add system candidates [Aug 18 10:34:25] DEBUG[14862] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:25] DEBUG[14862] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE add candidate: 159.65.48.104:16814, 2130706431 [Aug 18 10:34:25] DEBUG[14862] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:25] DEBUG[14862] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE add candidate: 10.131.0.10:16814, 2130706431 [Aug 18 10:34:25] DEBUG[14862] rtp_engine.c: RTP instance '0x7f0cac096810' is setup and ready to go [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) ICE stopped [Aug 18 10:34:25] DEBUG[14862] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:25] DEBUG[14862] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:25] DEBUG[14862] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:25] DEBUG[14862] res_rtp_asterisk.c: (0x7f0cac096810) RTCP setup on RTP instance [Aug 18 10:34:25] VERBOSE[14862] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:25] DEBUG[14862] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:25] DEBUG[14893] http.c: match request [ari/channels/213117] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14893] http.c: match request [ari/channels/213117] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213118, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213118': 0x7f0c7c094c80 destroyed [Aug 18 10:34:25] DEBUG[14893] http.c: match request [ari/channels/213117] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:601/channel:1629282865.523, detail: [Aug 18 10:34:25] DEBUG[14614] stasis.c: Creating topic. name: cache:600/channel:1629282865.522, detail: [Aug 18 10:34:25] DEBUG[14705] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14705] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:601/channel:1629282865.523': 0x7f0c300413e0 created [Aug 18 10:34:25] DEBUG[14614] stasis.c: Topic 'cache:600/channel:1629282865.522': 0x7f0c3007f570 created [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Finding handler for channels/213118 [Aug 18 10:34:25] DEBUG[14893] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Outgoing Call for 79821116882 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] VERBOSE[14892] chan_sip.c: Audio is at 13406 [Aug 18 10:34:25] DEBUG[14879] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c18049eb0' [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213115': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213115' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[14890] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Finding handler for channels/213117 [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:601/channel:1629282865.523, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:601/channel:1629282865.523': 0x7f0c300413e0 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.523, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.523': 0x7f0c300fba90 destroyed [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000098', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213115', '')] [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) RTP allocated port 10852 [Aug 18 10:34:25] DEBUG[14862] chan_sip.c: SIP call-id changed from '236f5bae10533ca0146105b1586ae542@127.0.1.1:5060' to '0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060' [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:382/channel:213115, detail: [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE creating session 0.0.0.0:10852 (10852) [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE create [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE add system candidates [Aug 18 10:34:25] DEBUG[14879] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:25] DEBUG[14879] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE add candidate: 159.65.48.104:10852, 2130706431 [Aug 18 10:34:25] DEBUG[14879] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:25] DEBUG[14879] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE add candidate: 10.131.0.10:10852, 2130706431 [Aug 18 10:34:25] DEBUG[14879] rtp_engine.c: RTP instance '0x7f0c18049eb0' is setup and ready to go [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) ICE stopped [Aug 18 10:34:25] DEBUG[14879] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:25] DEBUG[14879] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:25] DEBUG[14879] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:25] DEBUG[14879] res_rtp_asterisk.c: (0x7f0c18049eb0) RTCP setup on RTP instance [Aug 18 10:34:25] VERBOSE[14879] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:25] DEBUG[14879] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:25] DEBUG[14879] chan_sip.c: SIP call-id changed from '451071ca38cc328204fe5d2b2ad4fe1e@127.0.1.1:5060' to '761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060' [Aug 18 10:34:25] DEBUG[14879] stasis.c: Creating topic. name: channel:213183, detail: [Aug 18 10:34:25] DEBUG[14879] stasis.c: Topic 'channel:213183': 0x7f0c1805a800 created [Aug 18 10:34:25] DEBUG[14879] stasis.c: Creating topic. name: cache:602/channel:213183, detail: [Aug 18 10:34:25] DEBUG[14879] stasis.c: Topic 'cache:602/channel:213183': 0x7f0c1805b200 created [Aug 18 10:34:25] DEBUG[14896] http.c: HTTP opening session. Top level [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 From: ;tag=as671c682b To: ;tag=as53b926c5 Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as671c682b [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as53b926c5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 (Checking To) --From tag as671c682b --To-tag as53b926c5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Stopping retransmission on '2749fa7d41ec862f1556002a63546011@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (3) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (5) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2052644047 2052644047 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16832 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Initializing initreq for method INVITE - callid 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116883@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 [Aug 18 10:34:25] DEBUG[14862] stasis.c: Creating topic. name: channel:213175, detail: [Aug 18 10:34:25] DEBUG[14862] stasis.c: Topic 'channel:213175': 0x7f0cac03f050 created [Aug 18 10:34:25] DEBUG[14862] stasis.c: Creating topic. name: cache:603/channel:213175, detail: [Aug 18 10:34:25] DEBUG[14862] stasis.c: Topic 'cache:603/channel:213175': 0x7f0cac08ee40 created [Aug 18 10:34:25] VERBOSE[14892] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] VERBOSE[14892] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] VERBOSE[14892] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:382/channel:213115': 0x7f0c340411c0 destroyed [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.526, detail: [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Finding handler for 213117 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (2) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking channels create: Didn't match 213117 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Initializing initreq for method INVITE - callid 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116884@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 3 [ 52]: From: ;tag=as293daefc [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 6 [ 60]: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] VERBOSE[14892] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #40 [Aug 18 10:34:25] DEBUG[14892] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (5) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14893] res_ari.c: Checking channels externalMedia: Didn't match 213117 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116894@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051 Max-Forwards: 70 From: ;tag=as3e829f44 To: Contact: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1459261102 1459261102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:25] DEBUG[14893] res_ari.c: No explicit handler found for 213117. Using wildcard channelId. [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[14897] http.c: HTTP opening session. Top level [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '15d0975d5d60ded534cdf99c754106a5@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[14897] http.c: HTTP Request URI is /ari/channels/213116 [Aug 18 10:34:25] DEBUG[14897] http.c: match request [ari/channels/213116] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340b9d00) DTLS stop [Aug 18 10:34:25] DEBUG[14896] http.c: HTTP Request URI is /ari/channels/213115 [Aug 18 10:34:25] DEBUG[14897] http.c: match request [ari/channels/213116] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14897] http.c: match request [ari/channels/213116] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213116': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213116' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[14897] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.526': 0x7f0c300fba90 created [Aug 18 10:34:25] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Finding handler for channels/213116 [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Finding handler for channels [Aug 18 10:34:25] VERBOSE[14895] chan_sip.c: Audio is at 14482 [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 3 [ 52]: From: ;tag=as69c6d00c [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] VERBOSE[14892] dial.c: Called zvonobot/79821116884 [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:380/channel:213116, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:380/channel:213116': 0x7f0c280ead30 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213116, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213116': 0x7f0c280d2550 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213115, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213115': 0x7f0c340f1590 destroyed [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:604/channel:1629282865.526, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:604/channel:1629282865.526': 0x7f0c30169ba0 created [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] VERBOSE[14895] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] VERBOSE[14895] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] DEBUG[14602] res_stasis_recording.c: 1629282858.413: Sending record(212966_moZBbkKqaNszlKGcZcxmnwJIRmMzjXTL.wav) command [Aug 18 10:34:25] VERBOSE[14895] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Initializing initreq for method INVITE - callid 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116882@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 3 [ 52]: From: ;tag=as263c5372 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340b9d00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14602] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:25] DEBUG[14602] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340b9d00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c340b9d00) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c340b9d00' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4dd019b75f8af0fd2cada22c30b22fcb@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280cff50) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280cff50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280cff50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280cff50) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c280cff50' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0c2520) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0c2520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0c2520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c0c2520) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7c0c2520' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c11fbd0) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c11fbd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c11fbd0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c11fbd0) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3c11fbd0' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4871a5046aecc1db1b84f9484c4e7367@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280b2c40) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c280b2c40' [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350 Max-Forwards: 70 From: ;tag=as1f220605 To: ;tag=as0b424b33 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1ea6bc94", response="1600c32e40071d00c5ebdbecee32a250" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as1f220605 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0b424b33 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1ea6bc94", response="1600c32e40071d00c5ebdbecee32a250" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 (Checking From) --From tag as1f220605 --To-tag as0b424b33 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:25] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 6 [ 60]: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213026': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213026' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:604/channel:1629282865.526, detail: [Aug 18 10:34:25] DEBUG[14896] http.c: match request [ari/channels/213115] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:604/channel:1629282865.526': 0x7f0c30169ba0 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.526, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.526': 0x7f0c300fba90 destroyed [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000099', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213117', '')] [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14898] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14900] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14896] http.c: match request [ari/channels/213115] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 6 [ 60]: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] VERBOSE[14891] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Aug 18 10:34:25] DEBUG[14891] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Finding handler for 213118 [Aug 18 10:34:25] DEBUG[14900] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Finding handler for 213116 [Aug 18 10:34:25] DEBUG[14898] http.c: HTTP Request URI is /ari/channels/213026 [Aug 18 10:34:25] DEBUG[14900] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] VERBOSE[14895] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking channels create: Didn't match 213118 [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking channels create: Didn't match 213116 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #137 [Aug 18 10:34:25] DEBUG[14895] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14896] http.c: match request [ari/channels/213115] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14898] http.c: match request [ari/channels/213026] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:137/channel:213026, detail: [Aug 18 10:34:25] DEBUG[14896] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:137/channel:213026': 0x7f0c280b8290 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.527, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.527': 0x7f0c300fba90 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:605/channel:1629282865.527, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:605/channel:1629282865.527': 0x7f0c300413e0 created [Aug 18 10:34:25] DEBUG[14900] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14897] res_ari.c: Checking channels externalMedia: Didn't match 213116 [Aug 18 10:34:25] DEBUG[14894] res_ari.c: Checking channels externalMedia: Didn't match 213118 [Aug 18 10:34:25] DEBUG[14899] app.c: play_and_record: , /var/spool/asterisk/recording/212966_moZBbkKqaNszlKGcZcxmnwJIRmMzjXTL, 'wav' [Aug 18 10:34:25] DEBUG[14898] http.c: match request [ari/channels/213026] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14898] http.c: match request [ari/channels/213026] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14897] res_ari.c: No explicit handler found for 213116. Using wildcard channelId. [Aug 18 10:34:25] VERBOSE[14895] dial.c: Called zvonobot/79821116882 [Aug 18 10:34:25] DEBUG[14894] res_ari.c: No explicit handler found for 213118. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[14900] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213026, detail: [Aug 18 10:34:25] DEBUG[14898] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14899] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:25] DEBUG[14900] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Finding handler for channels/213115 [Aug 18 10:34:25] DEBUG[14477] chan_sip.c: Hangup call SIP/zvonobot-0000009b, SIP callid 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14477] res_rtp_asterisk.c: (0x7f0c84147390) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14477] res_rtp_asterisk.c: (0x7f0c84147390) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14477] channel.c: Channel 0x7f0c84094380 'SIP/zvonobot-0000009b' destroying [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Finding handler for bridges [Aug 18 10:34:25] DEBUG[14765] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Finding handler for bridges [Aug 18 10:34:25] DEBUG[14765] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213026': 0x7f0c280bc980 destroyed [Aug 18 10:34:25] VERBOSE[14891] dial.c: Called zvonobot/79821116883 [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Finding handler for channels/213026 [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] VERBOSE[14899] app.c: x=0, open writing: /var/spool/asterisk/recording/212966_moZBbkKqaNszlKGcZcxmnwJIRmMzjXTL format: wav, 0x7f0c940f5030 [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[13550] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-0000000e' [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:25] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14900] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:25] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:605/channel:1629282865.527, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:605/channel:1629282865.527': 0x7f0c300413e0 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.527, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.527': 0x7f0c300fba90 destroyed [Aug 18 10:34:25] DEBUG[14900] stasis.c: Creating topic. name: bridge:58483d29-982c-4e2a-bc1a-ca673af5c95a, detail: [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350;received=178.62.121.41 From: ;tag=as1f220605 To: ;tag=as0b424b33 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (3) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (5) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116892@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 Max-Forwards: 70 From: ;tag=as0a953bb4 To: Contact: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 772240936 772240936 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12378 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (1) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (1) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Session timer stopped: 25 - 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (1) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK45dad220 Max-Forwards: 70 From: ;tag=as5b70cf89 To: ;tag=as28933467 Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="335d9687", response="046207fa461d8a62d1d2bcf4883a256a" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK45dad220 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as5b70cf89 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as28933467 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="335d9687", response="046207fa461d8a62d1d2bcf4883a256a" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:25] DEBUG[14900] stasis.c: Topic 'bridge:58483d29-982c-4e2a-bc1a-ca673af5c95a': 0x7f0c900470e0 created [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[13550] channel.c: Deadlock avoided for write to channel 'SIP/zvonobot-0000000e' [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Finding handler for 213026 [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Setting 0x7f0c9809c220(SIP/zvonobot-0000000e) state from:0 to:1 [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking channels create: Didn't match 213026 [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009a', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213118', '')] [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 (Checking From) --From tag as5b70cf89 --To-tag as28933467 [Aug 18 10:34:25] DEBUG[14900] stasis.c: Creating topic. name: cache:606/bridge:58483d29-982c-4e2a-bc1a-ca673af5c95a, detail: [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14898] res_ari.c: Checking channels externalMedia: Didn't match 213026 [Aug 18 10:34:25] DEBUG[14898] res_ari.c: No explicit handler found for 213026. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:25] DEBUG[14900] stasis.c: Topic 'cache:606/bridge:58483d29-982c-4e2a-bc1a-ca673af5c95a': 0x7f0c90066960 created [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pulling 0x7f0c9809c220(SIP/zvonobot-0000000e) [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:25] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Finding handler for 213115 [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking channels create: Didn't match 213115 [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14896] res_ari.c: Checking channels externalMedia: Didn't match 213115 [Aug 18 10:34:25] DEBUG[14896] res_ari.c: No explicit handler found for 213115. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213122': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] VERBOSE[13550] bridge_channel.c: Channel SIP/zvonobot-0000000e left 'softmix' stasis-bridge [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.528, detail: [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c9809c220(SIP/zvonobot-0000000e) is leaving softmix technology [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Setting 0x7f0ca403e270(Announcer/ARI-0000002a;2) state from:0 to:2 [Aug 18 10:34:25] DEBUG[14900] bridge_native_rtp.c: Bridge '58483d29-982c-4e2a-bc1a-ca673af5c95a' can not use native RTP bridge as two channels are required [Aug 18 10:34:25] DEBUG[14904] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213122' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.528': 0x7f0c300fba90 created [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK45dad220;received=178.62.121.41 From: ;tag=as5b70cf89 To: ;tag=as28933467 Call-ID: 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14116] bridge_channel.c: Setting 0x7f0c20083a40(SIP/zvonobot-00000004) state from:0 to:1 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:607/channel:1629282865.528, detail: [Aug 18 10:34:25] DEBUG[14478] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:25] DEBUG[14479] chan_sip.c: Hangup call SIP/zvonobot-0000009c, SIP callid 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14900] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[14479] res_rtp_asterisk.c: (0x7f0c7804d100) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14479] res_rtp_asterisk.c: (0x7f0c7804d100) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14479] channel.c: Channel 0x7f0c78022bf0 'SIP/zvonobot-0000009c' destroying [Aug 18 10:34:25] DEBUG[14770] channel.c: Channel 0x7f0c08069930 'Recorder/ARI-00000048;1' allocated [Aug 18 10:34:25] DEBUG[14770] stasis.c: Creating topic. name: channel:1629282865.529, detail: [Aug 18 10:34:25] DEBUG[14770] stasis.c: Topic 'channel:1629282865.529': 0x7f0c0806b830 created [Aug 18 10:34:25] DEBUG[14770] stasis.c: Creating topic. name: cache:608/channel:1629282865.529, detail: [Aug 18 10:34:25] DEBUG[14770] stasis.c: Topic 'cache:608/channel:1629282865.529': 0x7f0c0806ba90 created [Aug 18 10:34:25] DEBUG[14478] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14904] http.c: HTTP Request URI is /ari/channels/213122 [Aug 18 10:34:25] DEBUG[14116] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pulling 0x7f0c20083a40(SIP/zvonobot-00000004) [Aug 18 10:34:25] VERBOSE[14116] bridge_channel.c: Channel SIP/zvonobot-00000004 left 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:25] DEBUG[14116] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c20083a40(SIP/zvonobot-00000004) is leaving simple_bridge technology [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[14900] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14116] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:25] DEBUG[14904] http.c: match request [ari/channels/213122] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14116] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[14116] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14116] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14116] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] DEBUG[14116] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:25] DEBUG[14900] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84147390) DTLS stop [Aug 18 10:34:25] DEBUG[14900] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:387/channel:213122, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:607/channel:1629282865.528': 0x7f0c300fb370 created [Aug 18 10:34:25] DEBUG[14904] http.c: match request [ari/channels/213122] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14900] bridge.c: Bridge 58483d29-982c-4e2a-bc1a-ca673af5c95a: calling simple_bridge technology constructor [Aug 18 10:34:25] DEBUG[14900] bridge.c: Bridge 58483d29-982c-4e2a-bc1a-ca673af5c95a: calling simple_bridge technology start [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84147390) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14904] http.c: match request [ari/channels/213122] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14904] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84147390) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c84147390) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[14116] bridge_channel.c: Bridge is returning 0x7f0c20083a40(SIP/zvonobot-00000004) to read format alaw [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c84147390' [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Finding handler for channels/213122 [Aug 18 10:34:25] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14116] channel.c: Channel SIP/zvonobot-00000004 setting read format path: ulaw -> alaw [Aug 18 10:34:25] DEBUG[14116] bridge_channel.c: Bridge is returning 0x7f0c20083a40(SIP/zvonobot-00000004) to write format alaw [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (3) INVITE - 5 [Aug 18 10:34:25] DEBUG[14116] channel.c: Channel SIP/zvonobot-00000004 setting write format path: alaw -> ulaw [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:25] DEBUG[14116] stasis/control.c: 212967, 9cefb3ad-33ea-4a52-96a1-42b677d6802c: Channel was departed from bridge [Aug 18 10:34:25] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:387/channel:213122': 0x7f0c84096460 destroyed [Aug 18 10:34:25] DEBUG[14905] http.c: HTTP opening session. Top level [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6267ms with no response [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14116] stasis/app.c: bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c': is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Session timer stopped: 84 - 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14631] channel.c: Channel 0x2c950d0 'SIP/zvonobot-000000ab' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[14905] http.c: HTTP Request URI is /ari/channels/212966/snoop?app=calls_0&spy=in [Aug 18 10:34:25] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[12888] stasis/control.c: 212967: Channel departing bridge [Aug 18 10:34:25] DEBUG[12888] bridge.c: Waiting for 0x7f0c20083a40(SIP/zvonobot-00000004) bridge thread to die. [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14900] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14900] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14116] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:25] DEBUG[12888] stasis/app.c: channel '212967': is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:607/channel:1629282865.528, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:607/channel:1629282865.528': 0x7f0c300fb370 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.528, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.528': 0x7f0c300fba90 destroyed [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000097', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213116', '')] [Aug 18 10:34:25] DEBUG[12888] channel.c: Channel 0x7f0c24011df0 'SIP/zvonobot-00000004' hanging up. Refs: 3 [Aug 18 10:34:25] WARNING[13678] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-0000001c;1 [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Finding handler for 213122 [Aug 18 10:34:25] DEBUG[14905] http.c: match request [ari/channels/212966/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking channels create: Didn't match 213122 [Aug 18 10:34:25] DEBUG[14905] http.c: match request [ari/channels/212966/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213122, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213122': 0x7f0c84109ae0 destroyed [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4;received=159.65.48.104 From: ;tag=as08a5ad00 To: ;tag=as331133ce Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27a12182" Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20fa3fb4;received=159.65.48.104 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as08a5ad00 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as331133ce [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27a12182" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 (Checking To) --From tag as08a5ad00 --To-tag as331133ce [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2cc23538293c1849651dca44558c8447@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6305ms with no response [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] DEBUG[14905] http.c: match request [ari/channels/212966/snoop] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[13550] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6'. Checking compatability for channels 'Announcer/ARI-0000002a;2' and 'Recorder/ARI-00000019;2' [Aug 18 10:34:25] DEBUG[13550] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as could not get details [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[13550] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] VERBOSE[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: switching from softmix technology to simple_bridge [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology constructor [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0ca403e270(Announcer/ARI-0000002a;2) to dummy bridge temporarily [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: moving 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) to dummy bridge temporarily [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0ca403e270(Announcer/ARI-0000002a;2) is leaving softmix technology (dummy) [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is leaving softmix technology (dummy) [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology stop [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0ca403e270(Announcer/ARI-0000002a;2) is joining simple_bridge technology [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Recorder/ARI-00000019;2 setting read format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Announcer/ARI-0000002a;2 setting write format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Announcer/ARI-0000002a;2 setting read format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is joining simple_bridge technology [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Recorder/ARI-00000019;2 setting read format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Announcer/ARI-0000002a;2 setting write format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Announcer/ARI-0000002a;2 setting read format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel Recorder/ARI-00000019;2 setting write format path: slin -> slin [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling simple_bridge technology start [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: deferring softmix technology destructor [Aug 18 10:34:25] DEBUG[13550] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: queueing action type:13 sub:1000 [Aug 18 10:34:25] DEBUG[14784] channel.c: Channel 0x7f0c2c0e17b0 'SIP/zvonobot-000000c9' allocated [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] DEBUG[14632] channel.c: Channel 0x7f0cac090e50 'SIP/zvonobot-000000aa' hanging up. Refs: 3 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (5) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116889@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 Max-Forwards: 70 From: ;tag=as1ac06673 To: Contact: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 722698992 722698992 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (2) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5129740f51f9292d29e823f263748e28@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5129740f51f9292d29e823f263748e28@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[14905] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7804d100) DTLS stop [Aug 18 10:34:25] DEBUG[14904] res_ari.c: Checking channels externalMedia: Didn't match 213122 [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14784] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7804d100) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14904] res_ari.c: No explicit handler found for 213122. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Finding handler for channels/212966/snoop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7804d100) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7804d100) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7804d100' [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.530, detail: [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Finding handler for 212966 [Aug 18 10:34:25] DEBUG[14784] res_stasis.c: calls_0: Subscribing to 213166 [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channels create: Didn't match 212966 [Aug 18 10:34:25] DEBUG[14784] stasis/app.c: Channel '213166' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Outgoing Call for 79821116874 [Aug 18 10:34:25] DEBUG[14784] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[14784] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channels externalMedia: Didn't match 212966 [Aug 18 10:34:25] DEBUG[14905] res_ari.c: No explicit handler found for 212966. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Finding handler for snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.530': 0x7f0c300b3d70 created [Aug 18 10:34:25] DEBUG[14905] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:609/channel:1629282865.530, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:609/channel:1629282865.530': 0x7f0c3010dc00 created [Aug 18 10:34:25] DEBUG[14162] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[14150] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pulling 0x7f0ca403e270(Announcer/ARI-0000002a;2) [Aug 18 10:34:25] DEBUG[14158] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: stopping mixing thread [Aug 18 10:34:25] VERBOSE[14150] bridge_channel.c: Channel Announcer/ARI-0000002a;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:25] DEBUG[14150] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0ca403e270(Announcer/ARI-0000002a;2) is leaving simple_bridge technology [Aug 18 10:34:25] DEBUG[14150] bridge_channel.c: Setting 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) state from:0 to:2 [Aug 18 10:34:25] DEBUG[14908] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14908] http.c: HTTP Request URI is /ari/channels/213119 [Aug 18 10:34:25] DEBUG[14908] http.c: match request [ari/channels/213119] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14908] http.c: match request [ari/channels/213119] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14150] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as two channels are required [Aug 18 10:34:25] DEBUG[14150] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[14150] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14908] http.c: match request [ari/channels/213119] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14150] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14150] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] DEBUG[14150] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 is already using the new technology. [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037;received=159.65.48.104 From: ;tag=as46d7f260 To: ;tag=as1d4dfdb7 Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fd37709" Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:25] DEBUG[14908] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0682c037;received=159.65.48.104 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as46d7f260 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1d4dfdb7 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[13624] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: pulling 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) [Aug 18 10:34:25] DEBUG[20534] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Finding handler for channels/213119 [Aug 18 10:34:25] VERBOSE[13624] bridge_channel.c: Channel Recorder/ARI-00000019;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:25] DEBUG[13624] bridge_channel.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: 0x7f0c7c00cdf0(Recorder/ARI-00000019;2) is leaving simple_bridge technology [Aug 18 10:34:25] VERBOSE[14906] chan_sip.c: Audio is at 19192 [Aug 18 10:34:25] DEBUG[20534] bridge_softmix.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6: Waiting for mixing thread to die. [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213119': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[13624] bridge_native_rtp.c: Bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6' can not use native RTP bridge as two channels are required [Aug 18 10:34:25] DEBUG[13624] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[13624] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[13624] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[13624] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] DEBUG[13624] bridge.c: Bridge d0f9af3e-7f00-4d11-8990-3d67ba7213d6 is already using the new technology. [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:609/channel:1629282865.530, detail: [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fd37709" [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:609/channel:1629282865.530': 0x7f0c3010dc00 destroyed [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[13624] channel.c: Channel 0x7f0c7c077520 'Recorder/ARI-00000019;2' hanging up. Refs: 2 [Aug 18 10:34:25] VERBOSE[14906] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Finding handler for 213119 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213119' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking channels create: Didn't match 213119 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 (Checking To) --From tag as46d7f260 --To-tag as1d4dfdb7 [Aug 18 10:34:25] DEBUG[14150] channel.c: Channel 0x7f0ca405f210 'Announcer/ARI-0000002a;2' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.530, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.530': 0x7f0c300b3d70 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:389/channel:213119, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:389/channel:213119': 0x7f0c780240d0 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213119, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213119': 0x7f0c78049400 destroyed [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 39f4d82472695eb235154ae4053a1efd@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14908] res_ari.c: Checking channels externalMedia: Didn't match 213119 [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Bridge is returning 0x7f0c9809c220(SIP/zvonobot-0000000e) to read format alaw [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14908] res_ari.c: No explicit handler found for 213119. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:54', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000043', '', 'AppDial2', '(Outgoing Line)', 27, 0, 'BUSY', 3, '', '213026', '')] [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:25] VERBOSE[14906] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6273ms with no response [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] VERBOSE[14906] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Initializing initreq for method INVITE - callid 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116874@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (2) INVITE - 5 [Aug 18 10:34:25] DEBUG[14639] channel.c: Channel 0x7f0c080f4050 'SIP/zvonobot-000000af' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 3 [ 52]: From: ;tag=as3f6b0566 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 6 [ 60]: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (2) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6276ms with no response [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel SIP/zvonobot-0000000e setting read format path: ulaw -> alaw [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] VERBOSE[14906] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14633] channel.c: Channel 0x7f0c24139530 'SIP/zvonobot-000000ac' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #48 [Aug 18 10:34:25] DEBUG[13550] bridge_channel.c: Bridge is returning 0x7f0c9809c220(SIP/zvonobot-0000000e) to write format alaw [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:25] DEBUG[14906] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[13550] channel.c: Channel SIP/zvonobot-0000000e setting write format path: alaw -> ulaw [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[13550] stasis/control.c: 212977, d0f9af3e-7f00-4d11-8990-3d67ba7213d6: Channel was departed from bridge [Aug 18 10:34:25] DEBUG[13550] stasis/app.c: bridge 'd0f9af3e-7f00-4d11-8990-3d67ba7213d6': is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14769] channel.c: Channel 0x7f0c100480d0 'SIP/zvonobot-000000c8' allocated [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14769] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[13550] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:25] DEBUG[12956] stasis/control.c: 212977: Channel departing bridge [Aug 18 10:34:25] DEBUG[12956] bridge.c: Waiting for 0x7f0c9809c220(SIP/zvonobot-0000000e) bridge thread to die. [Aug 18 10:34:25] DEBUG[12956] stasis/app.c: channel '212977': is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[12956] channel.c: Channel 0x7f0c9c00dcf0 'SIP/zvonobot-0000000e' hanging up. Refs: 3 [Aug 18 10:34:25] DEBUG[14782] channel.c: Channel 0x7f0c200e13c0 'SIP/zvonobot-000000ca' allocated [Aug 18 10:34:25] DEBUG[14484] chan_sip.c: Hangup call SIP/zvonobot-0000009f, SIP callid 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14484] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:25] DEBUG[14484] res_rtp_asterisk.c: (0x7f0c8006ad00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14484] res_rtp_asterisk.c: (0x7f0c8006ad00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14484] channel.c: Channel 0x7f0c8008bb10 'SIP/zvonobot-0000009f' destroying [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14782] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[14782] res_stasis.c: calls_0: Subscribing to 213168 [Aug 18 10:34:25] DEBUG[14782] stasis/app.c: Channel '213168' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Outgoing Call for 79821116872 [Aug 18 10:34:25] DEBUG[14782] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[14782] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] VERBOSE[14911] chan_sip.c: Audio is at 12396 [Aug 18 10:34:25] VERBOSE[14911] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] VERBOSE[14911] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] DEBUG[14890] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 31 instead [Aug 18 10:34:25] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.531, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.531': 0x7f0c300b3d70 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:610/channel:1629282865.531, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:610/channel:1629282865.531': 0x7f0c3010dc00 created [Aug 18 10:34:25] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 704, ms is 64 [Aug 18 10:34:25] VERBOSE[14911] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14769] res_stasis.c: calls_0: Subscribing to 213164 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14769] stasis/app.c: Channel '213164' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Outgoing Call for 79821116876 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 From: ;tag=as2ff9bb68 To: ;tag=as6dc6d37f Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] VERBOSE[14912] chan_sip.c: Audio is at 17132 [Aug 18 10:34:25] VERBOSE[14912] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Initializing initreq for method INVITE - callid 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14769] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116872@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 3 [ 52]: From: ;tag=as23425771 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 6 [ 60]: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] VERBOSE[14911] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #68 [Aug 18 10:34:25] DEBUG[14911] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:25] WARNING[13861] app.c: No audio available on Recorder/ARI-00000024;1?? [Aug 18 10:34:25] VERBOSE[13861] app.c: User hung up [Aug 18 10:34:25] DEBUG[13861] res_stasis_recording.c: 1629282840.195: Recording complete [Aug 18 10:34:25] DEBUG[13861] channel.c: Channel 0x7f0c9407a670 'Recorder/ARI-00000024;1' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[14730] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ff9bb68 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6dc6d37f [Aug 18 10:34:25] VERBOSE[14906] dial.c: Called zvonobot/79821116874 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:610/channel:1629282865.531, detail: [Aug 18 10:34:25] DEBUG[14769] http.c: HTTP closing session. Top level [Aug 18 10:34:25] VERBOSE[14912] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:610/channel:1629282865.531': 0x7f0c3010dc00 destroyed [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14786] channel.c: Channel 0x7f0c340d8d00 'SIP/zvonobot-000000cb' allocated [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] VERBOSE[14911] dial.c: Called zvonobot/79821116872 [Aug 18 10:34:25] DEBUG[14485] chan_sip.c: Hangup call SIP/zvonobot-0000009e, SIP callid 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14483] chan_sip.c: Hangup call SIP/zvonobot-0000009d, SIP callid 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14483] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:25] DEBUG[14483] res_rtp_asterisk.c: (0x7f0c940cb950) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14483] res_rtp_asterisk.c: (0x7f0c940cb950) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14483] channel.c: Channel 0x7f0c940ed7b0 'SIP/zvonobot-0000009d' destroying [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.531, detail: [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14786] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[14786] res_stasis.c: calls_0: Subscribing to 213169 [Aug 18 10:34:25] DEBUG[14786] stasis/app.c: Channel '213169' is 1 interested in calls_0 [Aug 18 10:34:25] DEBUG[13011] chan_sip.c: Hangup call SIP/zvonobot-00000017, SIP callid 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[13011] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[13011] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[13011] channel.c: Channel 0x7f0c3c020700 'SIP/zvonobot-00000017' destroying [Aug 18 10:34:25] DEBUG[14786] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[14620] channel.c: Channel 0x7f0c74046e10 'Announcer/ARI-00000042;2' allocated [Aug 18 10:34:25] DEBUG[14786] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14485] res_rtp_asterisk.c: (0x7f0c3803fd80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14485] res_rtp_asterisk.c: (0x7f0c3803fd80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] VERBOSE[14912] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14485] channel.c: Channel 0x7f0c38004d20 'SIP/zvonobot-0000009e' destroying [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[14620] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Outgoing Call for 79821116871 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.531': 0x7f0c300b3d70 destroyed [Aug 18 10:34:25] DEBUG[14620] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000042;1' [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14890] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Initializing initreq for method INVITE - callid 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14917] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c740684b0(Announcer/ARI-00000042;2) is joining [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009b', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213122', '')] [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116876@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 3 [ 52]: From: ;tag=as786713e2 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 6 [ 60]: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] VERBOSE[14912] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #134 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 (Checking To) --From tag as2ff9bb68 --To-tag as6dc6d37f [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Stopping retransmission on '11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14912] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213121': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213121' unsubscribed from calls_0 [Aug 18 10:34:25] VERBOSE[14915] chan_sip.c: Audio is at 12786 [Aug 18 10:34:25] VERBOSE[14912] dial.c: Called zvonobot/79821116876 [Aug 18 10:34:25] VERBOSE[14915] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:25] DEBUG[14917] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pushing 0x7f0c740684b0(Announcer/ARI-00000042;2) [Aug 18 10:34:25] DEBUG[14917] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:25] VERBOSE[14917] bridge_channel.c: Channel Announcer/ARI-00000042;2 joined 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:399/channel:213121, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:399/channel:213121': 0x7f0c800314b0 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213121, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213121': 0x7f0c80038640 destroyed [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.532, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.532': 0x7f0c300b3d70 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:611/channel:1629282865.532, detail: [Aug 18 10:34:25] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14927] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213123': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213123' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:611/channel:1629282865.532': 0x7f0c3010dc00 created [Aug 18 10:34:25] DEBUG[14927] http.c: HTTP Request URI is /ari/channels/213123 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:393/channel:213123, detail: [Aug 18 10:34:25] DEBUG[14927] http.c: match request [ari/channels/213123] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14927] http.c: match request [ari/channels/213123] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:393/channel:213123': 0x7f0c940cbea0 destroyed [Aug 18 10:34:25] DEBUG[14927] http.c: match request [ari/channels/213123] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:611/channel:1629282865.532, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:611/channel:1629282865.532': 0x7f0c3010dc00 destroyed [Aug 18 10:34:25] DEBUG[14927] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.532, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.532': 0x7f0c300b3d70 destroyed [Aug 18 10:34:25] DEBUG[14917] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009c', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213119', '')] [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (4) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14917] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Finding handler for channels/213123 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14917] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000004 - start 1629282822.143632 answer 1629282846.485110 end 1629282865.394064 dur 43.250 bill 18.908 dispo ANSWERED [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6465ms with no response [Aug 18 10:34:25] DEBUG[14917] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:25] DEBUG[14917] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000ab - start 1629282859.058217 answer 0.000000 end 1629282865.430329 dur 6.372 bill 1629282865.430 dispo NO ANSWER [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000000e - start 1629282824.115921 answer 1629282836.292665 end 1629282865.447720 dur 41.331 bill 29.155 dispo ANSWERED [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '212987': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '212987' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:30/channel:212987, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:30/channel:212987': 0x7f0c3c01b9e0 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:212987, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:212987': 0x7f0c3c023ca0 destroyed [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213123, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213123': 0x7f0c94029560 destroyed [Aug 18 10:34:25] DEBUG[14919] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000aa - start 1629282859.015167 answer 0.000000 end 1629282865.466518 dur 6.451 bill 1629282865.466 dispo NO ANSWER [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Finding handler for 213123 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking channels create: Didn't match 213123 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14927] res_ari.c: Checking channels externalMedia: Didn't match 213123 [Aug 18 10:34:25] DEBUG[14927] res_ari.c: No explicit handler found for 213123. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[14644] channel.c: Channel 0x7f0c1807ed80 'SIP/zvonobot-000000ae' hanging up. Refs: 3 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (1) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7a76a8977ef6105e1dac648529941a79@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8006ad00) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8006ad00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8006ad00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8006ad00) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8006ad00' [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6404ms with no response [Aug 18 10:34:25] WARNING[20585] chan_sip.c: Hanging up call 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000af - start 1629282859.238199 answer 0.000000 end 1629282865.590103 dur 6.351 bill 1629282865.590 dispo NO ANSWER [Aug 18 10:34:25] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000ac - start 1629282859.104555 answer 0.000000 end 1629282865.603540 dur 6.498 bill 1629282865.603 dispo NO ANSWER [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213120': is 0 interested in calls_0 [Aug 18 10:34:25] DEBUG[20620] stasis/app.c: channel '213120' unsubscribed from calls_0 [Aug 18 10:34:25] DEBUG[14917] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: cache:397/channel:213120, detail: [Aug 18 10:34:25] DEBUG[14917] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c740684b0(Announcer/ARI-00000042;2) is joining simple_bridge technology [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'cache:397/channel:213120': 0x7f0c38033170 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.533, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Destroying topic. name: channel:213120, detail: [Aug 18 10:34:25] DEBUG[20620] stasis.c: Topic 'channel:213120': 0x7f0c3808fbb0 destroyed [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[14928] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[14928] http.c: HTTP Request URI is /ari/channels/212987 [Aug 18 10:34:25] DEBUG[14928] http.c: match request [ari/channels/212987] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14928] http.c: match request [ari/channels/212987] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14928] http.c: match request [ari/channels/212987] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14928] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.533': 0x7f0c3005de60 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:612/channel:1629282865.533, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:612/channel:1629282865.533': 0x7f0c3003e1a0 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:612/channel:1629282865.533, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:612/channel:1629282865.533': 0x7f0c3003e1a0 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.533, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.533': 0x7f0c3005de60 destroyed [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009f', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213121', '')] [Aug 18 10:34:25] DEBUG[14646] channel.c: Channel 0x7f0cb407fe90 'SIP/zvonobot-000000ad' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[14620] res_stasis_playback.c: 1629282858.429: Sending play(sound:silence/2) command [Aug 18 10:34:25] VERBOSE[14915] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:25] VERBOSE[14915] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Initializing initreq for method INVITE - callid 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116871@178.62.121.41 SIP/2.0 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 3 [ 52]: From: ;tag=as0611ab7b [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 6 [ 60]: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:25 GMT [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:25] VERBOSE[14915] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #72 [Aug 18 10:34:25] DEBUG[14915] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[14620] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:25] DEBUG[14620] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14919] http.c: HTTP Request URI is /ari/channels/213121 [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788;received=159.65.48.104 From: ;tag=as2489799b To: ;tag=as2f32cc01 Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3fbca3fb" Content-Length: 0 <-------------> [Aug 18 10:34:25] DEBUG[14929] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Finding handler for channels/212987 [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Finding handler for 212987 [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking channels create: Didn't match 212987 [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14928] res_ari.c: Checking channels externalMedia: Didn't match 212987 [Aug 18 10:34:25] DEBUG[14928] res_ari.c: No explicit handler found for 212987. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:25] DEBUG[14918] http.c: HTTP opening session. Top level [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.534, detail: [Aug 18 10:34:25] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP got report of 100 bytes from 178.62.121.41:14927 [Aug 18 10:34:25] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 1088, ms is 88 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:25] DEBUG[14627] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:25] DEBUG[14627] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.534': 0x7f0c300d6ca0 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:613/channel:1629282865.534, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:613/channel:1629282865.534': 0x7f0c3003e1a0 created [Aug 18 10:34:25] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK14471788;received=159.65.48.104 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2489799b [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:613/channel:1629282865.534, detail: [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2f32cc01 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:613/channel:1629282865.534': 0x7f0c3003e1a0 destroyed [Aug 18 10:34:25] DEBUG[14929] http.c: HTTP Request URI is /ari/channels/213120 [Aug 18 10:34:25] DEBUG[14929] http.c: match request [ari/channels/213120] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.534, detail: [Aug 18 10:34:25] DEBUG[14919] http.c: match request [ari/channels/213121] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.534': 0x7f0c300d6ca0 destroyed [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:25] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:25] DEBUG[14919] http.c: match request [ari/channels/213121] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:25] DEBUG[14929] http.c: match request [ari/channels/213120] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14929] http.c: match request [ari/channels/213120] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14929] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Finding handler for channels/213120 [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009d', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213123', '')] [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.535, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.535': 0x7f0c3005de60 created [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: cache:614/channel:1629282865.535, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:614/channel:1629282865.535': 0x7f0c3003e1a0 created [Aug 18 10:34:25] VERBOSE[14915] dial.c: Called zvonobot/79821116871 [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: cache:614/channel:1629282865.535, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'cache:614/channel:1629282865.535': 0x7f0c3003e1a0 destroyed [Aug 18 10:34:25] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.535, detail: [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.535': 0x7f0c3005de60 destroyed [Aug 18 10:34:25] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000017', '', 'AppDial2', '(Outgoing Line)', 32, 0, 'BUSY', 3, '', '212987', '')] [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Finding handler for 213120 [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking channels create: Didn't match 213120 [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14929] res_ari.c: Checking channels externalMedia: Didn't match 213120 [Aug 18 10:34:25] DEBUG[14929] res_ari.c: No explicit handler found for 213120. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3fbca3fb" [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: = Looking for Call ID: 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 (Checking To) --From tag as2489799b --To-tag as2f32cc01 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (3) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (1) INVITE - 5 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060 [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3b3818513c0faf4f06bfff64213f0fb6@159.65.48.104:5060' Method: BYE [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS stop [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c40006350) ICE RTP transport deallocating [Aug 18 10:34:25] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282865.536, detail: [Aug 18 10:34:25] DEBUG[14918] http.c: HTTP Request URI is /ari/playbacks/6b5e6d30-80dd-436d-ba34-a3c872b140be [Aug 18 10:34:25] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c40006350' [Aug 18 10:34:25] DEBUG[14919] http.c: match request [ari/channels/213121] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14918] http.c: match request [ari/playbacks/6b5e6d30-80dd-436d-ba34-a3c872b140be] with handler [httpstatus] len 10 [Aug 18 10:34:25] DEBUG[14918] http.c: match request [ari/playbacks/6b5e6d30-80dd-436d-ba34-a3c872b140be] with handler [phoneprov] len 9 [Aug 18 10:34:25] DEBUG[14918] http.c: match request [ari/playbacks/6b5e6d30-80dd-436d-ba34-a3c872b140be] with handler [ari] len 3 [Aug 18 10:34:25] DEBUG[14918] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Finding handler for playbacks/6b5e6d30-80dd-436d-ba34-a3c872b140be [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Finding handler for playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Finding handler for 6b5e6d30-80dd-436d-ba34-a3c872b140be [Aug 18 10:34:25] DEBUG[14918] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:25] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:25] DEBUG[14918] res_ari.c: No explicit handler found for 6b5e6d30-80dd-436d-ba34-a3c872b140be. Using wildcard playbackId. [Aug 18 10:34:25] DEBUG[14919] http.c: Match made with [ari] [Aug 18 10:34:25] DEBUG[20585] chan_sip.c: Destroying SIP dialog 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Finding handler for channels/213121 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Finding handler for channels [Aug 18 10:34:25] DEBUG[14775] channel.c: Channel 0x7f0c1804ef10 'SIP/zvonobot-000000cc' allocated [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:25] DEBUG[14775] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:25] DEBUG[14162] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:25] DEBUG[14162] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:25] DEBUG[14162] channel.c: Channel Announcer/ARI-0000002a;1 setting write format path: slin -> slin [Aug 18 10:34:25] NOTICE[14162] res_stasis_playback.c: 1629282846.255: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:25] DEBUG[14162] channel.c: Channel 0x7f0ca4085510 'Announcer/ARI-0000002a;1' hanging up. Refs: 2 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:25] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060' Method: BYE [Aug 18 10:34:25] DEBUG[14918] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:25] DEBUG[20545] stasis.c: Topic 'channel:1629282865.536': 0x7f0c3003e1a0 created [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Finding handler for 213121 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking channels create: Didn't match 213121 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:25] DEBUG[14919] res_ari.c: Checking channels externalMedia: Didn't match 213121 [Aug 18 10:34:25] DEBUG[14919] res_ari.c: No explicit handler found for 213121. Using wildcard channelId. [Aug 18 10:34:25] DEBUG[14918] http.c: HTTP closing session. Top level [Aug 18 10:34:25] DEBUG[14931] channel.c: Channel Announcer/ARI-00000042;1 setting write format path: gsm -> slin [Aug 18 10:34:26] DEBUG[14773] channel.c: Channel 0x7f0c1c12d620 'SIP/zvonobot-000000cd' allocated [Aug 18 10:34:25] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS stop [Aug 18 10:34:26] DEBUG[14095] audiohook.c: Audiohook 0x7f0c3010f100 has stale audio in its factories. Flushing them both [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:25] DEBUG[14934] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8402e520) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8402e520' [Aug 18 10:34:26] DEBUG[14773] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Destroying SIP dialog 007f90413610c97471d9f37255f670d0@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14934] http.c: HTTP Request URI is /ari/channels/robot_212977 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '007f90413610c97471d9f37255f670d0@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:26] DEBUG[14931] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:26] VERBOSE[14931] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940cb950) DTLS stop [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940cb950) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940cb950) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940cb950) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c940cb950' [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7a20668b0696aba832a45ab76ac05073@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS stop [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c005640) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3c005640' [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Destroying SIP dialog 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3803fd80) DTLS stop [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3803fd80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[13428] res_rtp_asterisk.c: (0x7f0c74032f50) RTP 0x7f0c74038900 -- Received packet from 178.62.121.41:12664, dropping due to strict RTP protection. [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3803fd80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3803fd80) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[14630] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:26] DEBUG[14506] channel.c: Channel 0x7f0c1c0366a0 'Announcer/ARI-00000049;1' allocated [Aug 18 10:34:26] DEBUG[14506] stasis.c: Creating topic. name: channel:1629282866.537, detail: [Aug 18 10:34:26] DEBUG[14506] stasis.c: Topic 'channel:1629282866.537': 0x7f0c1c0a1aa0 created [Aug 18 10:34:26] DEBUG[14506] stasis.c: Creating topic. name: cache:616/channel:1629282866.537, detail: [Aug 18 10:34:26] DEBUG[14506] stasis.c: Topic 'cache:616/channel:1629282866.537': 0x7f0c1c053830 created [Aug 18 10:34:26] DEBUG[14506] channel.c: Channel 0x7f0c1c157030 'Announcer/ARI-00000049;2' allocated [Aug 18 10:34:26] DEBUG[14506] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:26] DEBUG[14934] http.c: match request [ari/channels/robot_212977] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3803fd80' [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:26] DEBUG[14506] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000049;1' [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: cache:615/channel:1629282865.536, detail: [Aug 18 10:34:26] DEBUG[14937] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) is joining [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:615/channel:1629282865.536': 0x7f0c300b3d70 created [Aug 18 10:34:26] DEBUG[14630] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14519] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP got report of 76 bytes from 178.62.121.41:12381 [Aug 18 10:34:26] DEBUG[14936] http.c: HTTP opening session. Top level [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14934] http.c: match request [ari/channels/robot_212977] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14934] http.c: match request [ari/channels/robot_212977] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14934] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14937] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pushing 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) [Aug 18 10:34:26] DEBUG[14626] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:26] DEBUG[14937] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:26] VERBOSE[14937] bridge_channel.c: Channel Announcer/ARI-00000049;2 joined 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:26] DEBUG[14626] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (1) INVITE - 5 [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (4) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as7e2e6628 Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1817551672 1817551672 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15352 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[14775] res_stasis.c: calls_0: Subscribing to 213165 [Aug 18 10:34:26] DEBUG[14775] stasis/app.c: Channel '213165' is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[14937] bridge.c: Chose bridge technology softmix [Aug 18 10:34:26] VERBOSE[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: switching from simple_bridge technology to softmix [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology constructor [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c18091350(SIP/zvonobot-00000013) to dummy bridge temporarily [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c940389d0(Recorder/ARI-00000024;2) to dummy bridge temporarily [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is leaving simple_bridge technology (dummy) [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:26] DEBUG[14775] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Outgoing Call for 79821116875 [Aug 18 10:34:26] DEBUG[14775] http.c: HTTP closing session. Top level [Aug 18 10:34:26] VERBOSE[14759] res_rtp_asterisk.c: 0x7f0c1c03e660 -- Strict RTP learning complete - Locking on source address 178.62.121.41:13598 [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology stop [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) is joining softmix technology [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Announcer/ARI-00000049;2: [Aug 18 10:34:26] DEBUG[14937] channel.c: Channel Announcer/ARI-00000049;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Announcer/ARI-00000049;2: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Announcer/ARI-00000049;2: Not in SFU mode [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining softmix technology [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: SIP/zvonobot-00000013: [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: SIP/zvonobot-00000013: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: SIP/zvonobot-00000013: Not in SFU mode [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is joining softmix technology [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Recorder/ARI-00000024;2: [Aug 18 10:34:26] DEBUG[14937] channel.c: Channel Recorder/ARI-00000024;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:26] DEBUG[14937] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Recorder/ARI-00000024;2: [Aug 18 10:34:26] DEBUG[14937] bridge_softmix.c: Recorder/ARI-00000024;2: Not in SFU mode [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology start [Aug 18 10:34:26] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology destructor [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7e2e6628 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1817551672 1817551672 IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15352 RTP/AVP 0 8 101 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:26] DEBUG[14936] http.c: HTTP Request URI is /ari/channels/213184?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116856&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14773] res_stasis.c: calls_0: Subscribing to 213167 [Aug 18 10:34:26] DEBUG[14773] stasis/app.c: Channel '213167' is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Finding handler for channels/robot_212977 [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:26] VERBOSE[14938] chan_sip.c: Audio is at 10394 [Aug 18 10:34:26] VERBOSE[14938] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14640] channel.c: Channel 0x7f0cac05f500 'Recorder/ARI-00000043;2' allocated [Aug 18 10:34:26] DEBUG[14640] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as7e2e6628 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6354ms with no response [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Hanging up call 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (1) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (6) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116898@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66601ae3 Max-Forwards: 70 From: ;tag=as35c0c7eb To: Contact: Call-ID: 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 144689560 144689560 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17048 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc;received=159.65.48.104 From: ;tag=as20932b4d To: ;tag=as2141137d Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d23c7e2" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as20932b4d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2141137d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d23c7e2" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 (Checking To) --From tag as20932b4d --To-tag as2141137d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580;received=159.65.48.104 From: ;tag=as54647e8b To: ;tag=as194fe9f0 Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22fa0926" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56951580;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54647e8b [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as194fe9f0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22fa0926" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 (Checking To) --From tag as54647e8b --To-tag as194fe9f0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1166f54c75ddfc7d3bab0bef30395ba9@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 Max-Forwards: 70 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as20bcc5bd [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as31e40966 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="34e58402", response="7311b45c98cebd99bc55d76c97e3d398" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 (Checking From) --From tag as20bcc5bd --To-tag as31e40966 [Aug 18 10:34:26] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting AST_TRANSPORT_UNKNOWN with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK132f2e09;received=178.62.121.41 From: ;tag=as20bcc5bd To: ;tag=as31e40966 Call-ID: 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 6a169cdc3aa091076d4920f31d70c805@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 593 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 From: ;tag=as2ff9bb68 To: ;tag=as6dc6d37f Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ff9bb68 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6dc6d37f [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Finding handler for robot_212977 [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking channels create: Didn't match robot_212977 [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14934] res_ari.c: Checking channels externalMedia: Didn't match robot_212977 [Aug 18 10:34:26] DEBUG[14934] res_ari.c: No explicit handler found for robot_212977. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 (Checking To) --From tag as2ff9bb68 --To-tag as6dc6d37f [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116911@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776 Max-Forwards: 70 From: ;tag=as2ff9bb68 To: Contact: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5 Max-Forwards: 70 From: ;tag=as6847ab41 To: ;tag=as0d63cc42 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="5cc7861d", response="2ddbce0d9077c405df851db6d16b3c6e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6847ab41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0d63cc42 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="5cc7861d", response="2ddbce0d9077c405df851db6d16b3c6e" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 (Checking From) --From tag as6847ab41 --To-tag as0d63cc42 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:26] VERBOSE[14938] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:26] DEBUG[14939] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: starting mixing thread [Aug 18 10:34:26] DEBUG[14660] channel.c: Channel 0x7f0c2c09fc70 'SIP/zvonobot-000000b0' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[13704] audiohook.c: Audiohook 0x7f0c8805b620 has stale audio in its factories. Flushing them both [Aug 18 10:34:26] DEBUG[14941] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0cac01a9f0(Recorder/ARI-00000043;2) is joining [Aug 18 10:34:26] DEBUG[14506] res_stasis_playback.c: 1629282862.480: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:26] DEBUG[14506] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:26] DEBUG[14506] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP audio difference is 51448, ms is 6451 [Aug 18 10:34:26] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14773] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:26] DEBUG[14773] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14890] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14941] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: pushing 0x7f0cac01a9f0(Recorder/ARI-00000043;2) [Aug 18 10:34:26] DEBUG[14936] http.c: match request [ari/channels/213184] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14936] http.c: match request [ari/channels/213184] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14936] http.c: match request [ari/channels/213184] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14936] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14936] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: cache:615/channel:1629282865.536, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:615/channel:1629282865.536': 0x7f0c300b3d70 destroyed [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282865.536, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282865.536': 0x7f0c3003e1a0 destroyed [Aug 18 10:34:26] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:15', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000009e', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213120', '')] [Aug 18 10:34:26] DEBUG[14941] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:26] VERBOSE[14941] bridge_channel.c: Channel Recorder/ARI-00000043;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:26] DEBUG[14759] res_rtp_asterisk.c: (0x7f0c1c00b650) RTCP got report of 76 bytes from 178.62.121.41:13599 [Aug 18 10:34:26] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14941] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10'. Checking compatability for channels 'SIP/zvonobot-00000014' and 'Recorder/ARI-00000043;2' [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Finding handler for channels/213184 [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Finding handler for 213184 [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking channels create: Didn't match 213184 [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14936] res_ari.c: Checking channels externalMedia: Didn't match 213184 [Aug 18 10:34:26] DEBUG[14936] res_ari.c: No explicit handler found for 213184. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[14941] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' can not use native RTP bridge as could not get details [Aug 18 10:34:26] DEBUG[13379] res_rtp_asterisk.c: (0x7f0ca406c1b0) RTP audio difference is 864, ms is 74 [Aug 18 10:34:26] DEBUG[14943] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14038] res_rtp_asterisk.c: (0x7f0ca003db80) RTP audio difference is 768, ms is 68 [Aug 18 10:34:26] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP no remote address on instance, so dropping frame [Aug 18 10:34:26] DEBUG[14941] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[14955] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP no remote address on instance, so dropping frame [Aug 18 10:34:26] DEBUG[14941] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 672, ms is 62 [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14955] http.c: HTTP Request URI is /ari/channels/213187?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116853&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14943] http.c: HTTP Request URI is /ari/channels/213185?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116855&callerId=74950493843 [Aug 18 10:34:26] VERBOSE[14938] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Initializing initreq for method INVITE - callid 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116875@178.62.121.41 SIP/2.0 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 3 [ 52]: From: ;tag=as7e29cf80 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 6 [ 60]: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:26 GMT [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] VERBOSE[14938] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Aug 18 10:34:26] DEBUG[14938] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[14938] dial.c: Called zvonobot/79821116875 [Aug 18 10:34:26] DEBUG[14941] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14941] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[14943] http.c: match request [ari/channels/213185] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14943] http.c: match request [ari/channels/213185] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14943] http.c: match request [ari/channels/213185] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14943] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14943] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Finding handler for channels/213185 [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[13441] res_rtp_asterisk.c: (0x7f0c8c00f7e0) RTP no remote address on instance, so dropping frame [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5;received=178.62.121.41 From: ;tag=as6847ab41 To: ;tag=as0d63cc42 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (5) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6405ms with no response [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Hanging up call 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Session timer stopped: 20 - 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14659] channel.c: Channel 0x7f0c280eeac0 'SIP/zvonobot-000000b1' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14955] http.c: match request [ari/channels/213187] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14941] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10 is already using the new technology. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (4) INVITE - 5 [Aug 18 10:34:26] DEBUG[14947] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14941] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0cac01a9f0(Recorder/ARI-00000043;2) is joining simple_bridge technology [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Outgoing Call for 79821116873 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:26] VERBOSE[14940] chan_sip.c: Audio is at 13196 [Aug 18 10:34:26] VERBOSE[14940] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:26] VERBOSE[14940] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:26] VERBOSE[14940] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Initializing initreq for method INVITE - callid 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116873@178.62.121.41 SIP/2.0 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 3 [ 52]: From: ;tag=as109bf1f8 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 6 [ 60]: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:26 GMT [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] VERBOSE[14940] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #110 [Aug 18 10:34:26] DEBUG[14940] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[14940] dial.c: Called zvonobot/79821116873 [Aug 18 10:34:26] DEBUG[14956] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000ae - start 1629282859.362371 answer 0.000000 end 1629282865.882041 dur 6.519 bill 1629282865.882 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000ad - start 1629282859.417642 answer 0.000000 end 1629282865.896940 dur 6.479 bill 1629282865.896 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14941] channel.c: Channel Recorder/ARI-00000043;2 setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Setting 0x7f0c88050c90(SIP/zvonobot-00000020) state from:0 to:1 [Aug 18 10:34:26] DEBUG[14956] http.c: HTTP Request URI is /ari/channels/213190?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116850&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14941] channel.c: Channel SIP/zvonobot-00000014 setting write format path: slin -> alaw [Aug 18 10:34:26] DEBUG[14956] http.c: match request [ari/channels/213190] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14956] http.c: match request [ari/channels/213190] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pulling 0x7f0c88050c90(SIP/zvonobot-00000020) [Aug 18 10:34:26] VERBOSE[13441] bridge_channel.c: Channel SIP/zvonobot-00000020 left 'softmix' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c88050c90(SIP/zvonobot-00000020) is leaving softmix technology [Aug 18 10:34:26] DEBUG[14794] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:26] DEBUG[14794] http.c: HTTP closing session. Top level [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK36ef83da Max-Forwards: 70 From: ;tag=as228e10c3 To: ;tag=as5e4952fc Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="49e49fba", response="c6b68d7eb6af42cbe6c3f24d3f851bf7" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK36ef83da [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as228e10c3 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as5e4952fc [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="49e49fba", response="c6b68d7eb6af42cbe6c3f24d3f851bf7" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 (Checking From) --From tag as228e10c3 --To-tag as5e4952fc [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14939] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Setting 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) state from:0 to:2 [Aug 18 10:34:26] DEBUG[14947] http.c: HTTP Request URI is /ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:26] DEBUG[14950] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[13183] chan_sip.c: Hangup call SIP/zvonobot-0000002b, SIP callid 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14955] http.c: match request [ari/channels/213187] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14950] http.c: HTTP Request URI is /ari/channels/213186?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116854&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14957] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[13183] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[13183] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] VERBOSE[13183] chan_sip.c: Scheduling destruction of SIP dialog '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' in 6400 ms (Method: INVITE) [Aug 18 10:34:26] DEBUG[13183] chan_sip.c: Strict routing enforced for session 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[13183] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:26] DEBUG[13183] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[13183] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] DEBUG[13762] bridge_channel.c: Setting 0x7f0c940356f0(Snoop/213008-0000000a) state from:0 to:1 [Aug 18 10:34:26] DEBUG[14957] http.c: HTTP Request URI is /ari/channels/213188?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116852&callerId=74950493843 [Aug 18 10:34:26] VERBOSE[13183] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[13183] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[13183] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #24 [Aug 18 10:34:26] DEBUG[13183] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[13441] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804'. Checking compatability for channels 'Announcer/ARI-0000001b;2' and 'Recorder/ARI-00000011;2' [Aug 18 10:34:26] DEBUG[13762] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: pulling 0x7f0c940356f0(Snoop/213008-0000000a) [Aug 18 10:34:26] VERBOSE[13762] bridge_channel.c: Channel Snoop/213008-0000000a left 'simple_bridge' stasis-bridge [Aug 18 10:34:26] DEBUG[13762] bridge_channel.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: 0x7f0c940356f0(Snoop/213008-0000000a) is leaving simple_bridge technology [Aug 18 10:34:26] DEBUG[13762] bridge_native_rtp.c: Bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[13762] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[13762] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13762] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13762] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[13762] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a is already using the new technology. [Aug 18 10:34:26] DEBUG[13762] bridge_channel.c: Bridge is returning 0x7f0c940356f0(Snoop/213008-0000000a) to read format slin [Aug 18 10:34:26] DEBUG[13762] channel.c: Channel Snoop/213008-0000000a setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[13762] bridge_channel.c: Bridge is returning 0x7f0c940356f0(Snoop/213008-0000000a) to write format slin [Aug 18 10:34:26] DEBUG[13762] channel.c: Channel Snoop/213008-0000000a setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[13762] stasis/control.c: 1629282839.182, e0573cd4-75f6-4425-a1e4-83029f01aa9a: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[13762] stasis/app.c: bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a': is 2 interested in calls_0 [Aug 18 10:34:26] DEBUG[13762] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[14941] channel.c: Channel SIP/zvonobot-00000014 setting read format path: alaw -> slin [Aug 18 10:34:26] DEBUG[14956] http.c: match request [ari/channels/213190] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[13441] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as could not get details [Aug 18 10:34:26] DEBUG[13686] stasis/control.c: 1629282839.182: Channel departing bridge [Aug 18 10:34:26] DEBUG[14959] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14941] channel.c: Channel Recorder/ARI-00000043;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[13686] bridge.c: Waiting for 0x7f0c940356f0(Snoop/213008-0000000a) bridge thread to die. [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13441] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[14956] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[13686] stasis/app.c: channel '1629282839.182': is 0 interested in calls_0 [Aug 18 10:34:26] DEBUG[13686] stasis/app.c: channel '1629282839.182' unsubscribed from calls_0 [Aug 18 10:34:26] DEBUG[13686] channel.c: Channel 0x7f0ca0076bd0 'Snoop/213008-0000000a' hanging up. Refs: 3 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:26] VERBOSE[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: switching from softmix technology to simple_bridge [Aug 18 10:34:26] DEBUG[14959] http.c: HTTP Request URI is /ari/channels/213192?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116848&callerId=74950493843 [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology constructor [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) to dummy bridge temporarily [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:26] DEBUG[14957] http.c: match request [ari/channels/213188] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14957] http.c: match request [ari/channels/213188] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[14611] bridge_channel.c: Setting 0x7f0c900b05e0(SIP/zvonobot-0000003a) state from:0 to:1 [Aug 18 10:34:26] DEBUG[14611] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: pulling 0x7f0c900b05e0(SIP/zvonobot-0000003a) [Aug 18 10:34:26] VERBOSE[14611] bridge_channel.c: Channel SIP/zvonobot-0000003a left 'simple_bridge' stasis-bridge <5c24e2ba-8671-4745-b349-4500db0d3cb5> [Aug 18 10:34:26] DEBUG[14611] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: 0x7f0c900b05e0(SIP/zvonobot-0000003a) is leaving simple_bridge technology [Aug 18 10:34:26] DEBUG[14611] bridge_native_rtp.c: Bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[14611] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[14611] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[14611] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[14611] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[14611] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5 is already using the new technology. [Aug 18 10:34:26] DEBUG[14611] stasis/control.c: 213021, 5c24e2ba-8671-4745-b349-4500db0d3cb5: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[14611] stasis/app.c: bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5': is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: moving 0x7f0c180a6470(Recorder/ARI-00000011;2) to dummy bridge temporarily [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) is leaving softmix technology (dummy) [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is leaving softmix technology (dummy) [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology stop [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK36ef83da;received=178.62.121.41 From: ;tag=as228e10c3 To: ;tag=as5e4952fc Call-ID: 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:26] DEBUG[13289] stasis/control.c: 213021: Channel departing bridge [Aug 18 10:34:26] DEBUG[14611] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[13289] bridge.c: Waiting for 0x7f0c900b05e0(SIP/zvonobot-0000003a) bridge thread to die. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6299ms with no response [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Hanging up call 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Session timer stopped: 19 - 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6272ms with no response [Aug 18 10:34:26] WARNING[20585] chan_sip.c: Hanging up call 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14680] channel.c: Channel 0x7f0c200b11b0 'SIP/zvonobot-000000b3' hanging up. Refs: 3 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Session timer stopped: 127 - 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 From: ;tag=as4e77dae5 To: ;tag=as54b5330a Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 309728419 309728419 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16904 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31437222;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4e77dae5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as54b5330a [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 309728419 309728419 IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16904 RTP/AVP 0 8 101 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 (Checking To) --From tag as4e77dae5 --To-tag as54b5330a [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '7804b47921ede33542c911be5c80d598@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Strict routing enforced for session 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117013@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cbced8 Max-Forwards: 70 From: ;tag=as4e77dae5 To: ;tag=as54b5330a Contact: Call-ID: 7804b47921ede33542c911be5c80d598@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14939] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: Allocating new SIP dialog for 6611a0173d46ce34252967f8571330bd@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14950] http.c: match request [ari/channels/213186] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14956] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14955] http.c: match request [ari/channels/213187] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14950] http.c: match request [ari/channels/213186] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14674] channel.c: Channel 0x7f0c300fa180 'SIP/zvonobot-000000b2' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14947] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) is joining simple_bridge technology [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:26] DEBUG[14950] http.c: match request [ari/channels/213186] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14947] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14762] res_rtp_asterisk.c: (0x7f0c90008240) RTCP got report of 76 bytes from 178.62.121.41:12819 [Aug 18 10:34:26] VERBOSE[14762] res_rtp_asterisk.c: 0x7f0c9000c310 -- Strict RTP learning complete - Locking on source address 178.62.121.41:12818 [Aug 18 10:34:26] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP got report of 100 bytes from 178.62.121.41:10695 [Aug 18 10:34:26] DEBUG[14947] http.c: match request [ari/bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14955] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14957] http.c: match request [ari/channels/213188] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14959] http.c: match request [ari/channels/213192] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14947] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b0 - start 1629282859.696847 answer 0.000000 end 1629282866.094939 dur 6.398 bill 1629282866.094 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[14950] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Recorder/ARI-00000011;2 setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Announcer/ARI-0000001b;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Announcer/ARI-0000001b;2 setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[13704] res_rtp_asterisk.c: (0x7f0c1c00bc40) RTP audio difference is 1024, ms is 148 [Aug 18 10:34:26] DEBUG[13289] stasis/app.c: channel '213021': is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[13289] channel.c: Channel 0x7f0c700595e0 'SIP/zvonobot-0000003a' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14960] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14957] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[14960] http.c: HTTP Request URI is /ari/channels/213189?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116851&callerId=74950493843 [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is joining simple_bridge technology [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Recorder/ARI-00000011;2 setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b1 - start 1629282859.672357 answer 0.000000 end 1629282866.166241 dur 6.493 bill 1629282866.166 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Announcer/ARI-0000001b;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Finding handler for channels/213190 [Aug 18 10:34:26] DEBUG[14960] http.c: match request [ari/channels/213189] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14936] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2c0d05a0' [Aug 18 10:34:26] DEBUG[14957] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Announcer/ARI-0000001b;2 setting read format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel Recorder/ARI-00000011;2 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14959] http.c: match request [ari/channels/213192] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling simple_bridge technology start [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: deferring softmix technology destructor [Aug 18 10:34:26] DEBUG[13441] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: queueing action type:13 sub:1000 [Aug 18 10:34:26] DEBUG[14955] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14960] http.c: match request [ari/channels/213189] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14959] http.c: match request [ari/channels/213192] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:26] DEBUG[14950] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Finding handler for bridges/fbfe71c6-df7c-4b8c-8b66-37207aaf9846/play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Finding handler for bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Finding handler for fbfe71c6-df7c-4b8c-8b66-37207aaf9846 [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14947] res_ari.c: No explicit handler found for fbfe71c6-df7c-4b8c-8b66-37207aaf9846. Using wildcard bridgeId. [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Finding handler for play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:26] DEBUG[14947] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Finding handler for 213190 [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking channels create: Didn't match 213190 [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14956] res_ari.c: Checking channels externalMedia: Didn't match 213190 [Aug 18 10:34:26] DEBUG[14956] res_ari.c: No explicit handler found for 213190. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14962] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14640] res_stasis_recording.c: 1629282859.434: Sending record(212984_MMwrJXPbQmVTxOFiVLbLPNOJkQFoMEGA.wav) command [Aug 18 10:34:26] DEBUG[14959] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14959] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) RTP allocated port 12150 [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE creating session 0.0.0.0:12150 (12150) [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE create [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE add system candidates [Aug 18 10:34:26] DEBUG[14936] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14936] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE add candidate: 159.65.48.104:12150, 2130706431 [Aug 18 10:34:26] DEBUG[14936] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14936] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE add candidate: 10.131.0.10:12150, 2130706431 [Aug 18 10:34:26] DEBUG[14936] rtp_engine.c: RTP instance '0x7f0c2c0d05a0' is setup and ready to go [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) ICE stopped [Aug 18 10:34:26] DEBUG[14936] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14936] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14640] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:26] DEBUG[14640] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Finding handler for 213185 [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking channels create: Didn't match 213185 [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14943] res_ari.c: Checking channels externalMedia: Didn't match 213185 [Aug 18 10:34:26] DEBUG[14943] res_ari.c: No explicit handler found for 213185. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Finding handler for channels/213187 [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Finding handler for 213187 [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking channels create: Didn't match 213187 [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14955] res_ari.c: Checking channels externalMedia: Didn't match 213187 [Aug 18 10:34:26] DEBUG[14955] res_ari.c: No explicit handler found for 213187. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14962] http.c: HTTP Request URI is /ari/channels/213193?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116847&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14965] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14965] http.c: HTTP Request URI is /ari/channels/213191?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116849&callerId=74950493843 [Aug 18 10:34:26] DEBUG[14965] http.c: match request [ari/channels/213191] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14964] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14960] http.c: match request [ari/channels/213189] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Finding handler for channels/213192 [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Finding handler for channels/213188 [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14963] app.c: play_and_record: , /var/spool/asterisk/recording/212984_MMwrJXPbQmVTxOFiVLbLPNOJkQFoMEGA, 'wav' [Aug 18 10:34:26] DEBUG[14963] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14939] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14960] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14962] http.c: match request [ari/channels/213193] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14962] http.c: match request [ari/channels/213193] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14962] http.c: match request [ari/channels/213193] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14936] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14936] res_rtp_asterisk.c: (0x7f0c2c0d05a0) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14936] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14960] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000020 - start 1629282828.056374 answer 1629282835.080593 end 1629282866.209088 dur 38.152 bill 31.128 dispo ANSWERED [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14936] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14962] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14946] channel.c: Channel Announcer/ARI-00000049;1 setting write format path: gsm -> slin [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Finding handler for channels/213186 [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] VERBOSE[14963] app.c: x=0, open writing: /var/spool/asterisk/recording/212984_MMwrJXPbQmVTxOFiVLbLPNOJkQFoMEGA format: wav, 0x7f0c90068bc0 [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Finding handler for channels/213189 [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14946] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:26] VERBOSE[14946] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:26] DEBUG[14965] http.c: match request [ari/channels/213191] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14965] http.c: match request [ari/channels/213191] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14965] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14965] http.c: HTTP consuming request body [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Finding handler for channels/213191 [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Finding handler for channels/213193 [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Finding handler for 213192 [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking channels create: Didn't match 213192 [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14959] res_ari.c: Checking channels externalMedia: Didn't match 213192 [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Finding handler for 213193 [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking channels create: Didn't match 213193 [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14962] res_ari.c: Checking channels externalMedia: Didn't match 213193 [Aug 18 10:34:26] DEBUG[14962] res_ari.c: No explicit handler found for 213193. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14959] res_ari.c: No explicit handler found for 213192. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 465 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 465 [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Finding handler for 213189 [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking channels create: Didn't match 213189 [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14960] res_ari.c: Checking channels externalMedia: Didn't match 213189 [Aug 18 10:34:26] DEBUG[14960] res_ari.c: No explicit handler found for 213189. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14964] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Finding handler for 213186 [Aug 18 10:34:26] DEBUG[14730] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking channels create: Didn't match 213186 [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Finding handler for 213191 [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking channels create: Didn't match 213191 [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14965] res_ari.c: Checking channels externalMedia: Didn't match 213191 [Aug 18 10:34:26] DEBUG[14965] res_ari.c: No explicit handler found for 213191. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14950] res_ari.c: Checking channels externalMedia: Didn't match 213186 [Aug 18 10:34:26] DEBUG[14950] res_ari.c: No explicit handler found for 213186. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14856] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:26] DEBUG[14856] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:26] DEBUG[14964] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14057] res_rtp_asterisk.c: (0x7f0c240f8a30) RTP audio difference is 640, ms is 60 [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Finding handler for 213188 [Aug 18 10:34:26] DEBUG[14856] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:26] DEBUG[14856] channel.c: Channel Announcer/ARI-0000003d;1 setting write format path: slin -> slin [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Bridge is returning 0x7f0c88050c90(SIP/zvonobot-00000020) to read format alaw [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel SIP/zvonobot-00000020 setting read format path: alaw -> alaw [Aug 18 10:34:26] DEBUG[13441] bridge_channel.c: Bridge is returning 0x7f0c88050c90(SIP/zvonobot-00000020) to write format alaw [Aug 18 10:34:26] DEBUG[13441] channel.c: Channel SIP/zvonobot-00000020 setting write format path: alaw -> alaw [Aug 18 10:34:26] DEBUG[13441] stasis/control.c: 212995, 0aaea81d-67a8-499e-9e08-2fb745e40804: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[13441] stasis/app.c: bridge '0aaea81d-67a8-499e-9e08-2fb745e40804': is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[13441] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:26] DEBUG[13676] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: stopping mixing thread [Aug 18 10:34:26] DEBUG[13672] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pulling 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) [Aug 18 10:34:26] VERBOSE[13672] bridge_channel.c: Channel Announcer/ARI-0000001b;2 left 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:34:26] DEBUG[13672] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c7005bfb0(Announcer/ARI-0000001b;2) is leaving simple_bridge technology [Aug 18 10:34:26] DEBUG[13672] bridge_channel.c: Setting 0x7f0c180a6470(Recorder/ARI-00000011;2) state from:0 to:2 [Aug 18 10:34:26] DEBUG[13672] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[13672] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[13672] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13672] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13672] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[13672] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 is already using the new technology. [Aug 18 10:34:26] DEBUG[13443] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: pulling 0x7f0c180a6470(Recorder/ARI-00000011;2) [Aug 18 10:34:26] VERBOSE[13443] bridge_channel.c: Channel Recorder/ARI-00000011;2 left 'simple_bridge' stasis-bridge <0aaea81d-67a8-499e-9e08-2fb745e40804> [Aug 18 10:34:26] DEBUG[13443] bridge_channel.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: 0x7f0c180a6470(Recorder/ARI-00000011;2) is leaving simple_bridge technology [Aug 18 10:34:26] DEBUG[13443] bridge_native_rtp.c: Bridge '0aaea81d-67a8-499e-9e08-2fb745e40804' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[13443] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[13443] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13443] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[13443] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[13443] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804 is already using the new technology. [Aug 18 10:34:26] DEBUG[13443] channel.c: Channel 0x7f0c18087070 'Recorder/ARI-00000011;2' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14856] channel.c: Channel 0x7f0c8c10bad0 'Announcer/ARI-0000003d;1' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking channels create: Didn't match 213188 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14957] res_ari.c: Checking channels externalMedia: Didn't match 213188 [Aug 18 10:34:26] DEBUG[14957] res_ari.c: No explicit handler found for 213188. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[20534] bridge.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:26] DEBUG[20534] bridge_softmix.c: Bridge 0aaea81d-67a8-499e-9e08-2fb745e40804: Waiting for mixing thread to die. [Aug 18 10:34:26] DEBUG[13672] channel.c: Channel 0x7f0c7007bb40 'Announcer/ARI-0000001b;2' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[13114] stasis/control.c: 212995: Channel departing bridge [Aug 18 10:34:26] DEBUG[13114] bridge.c: Waiting for 0x7f0c88050c90(SIP/zvonobot-00000020) bridge thread to die. [Aug 18 10:34:26] DEBUG[13114] stasis/app.c: channel '212995': is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[13114] channel.c: Channel 0x7f0c8c0178b0 'SIP/zvonobot-00000020' hanging up. Refs: 3 [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000003a - start 1629282832.327994 answer 1629282857.933267 end 1629282866.226079 dur 33.898 bill 8.292 dispo ANSWERED [Aug 18 10:34:26] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14839] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14964] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14966] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b3 - start 1629282859.891421 answer 0.000000 end 1629282866.230396 dur 6.338 bill 1629282866.230 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b2 - start 1629282859.838665 answer 0.000000 end 1629282866.234144 dur 6.395 bill 1629282866.234 dispo NO ANSWER [Aug 18 10:34:26] DEBUG[14966] http.c: HTTP Request URI is /ari/playbacks/b52a589b-36b3-45e3-bd52-bf3ffebcf94d [Aug 18 10:34:26] DEBUG[14939] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:26] DEBUG[14649] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:26] DEBUG[14649] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14964] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14936] chan_sip.c: SIP call-id changed from '6611a0173d46ce34252967f8571330bd@127.0.1.1:5060' to '08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14936] stasis.c: Creating topic. name: channel:213184, detail: [Aug 18 10:34:26] DEBUG[14936] stasis.c: Topic 'channel:213184': 0x7f0c2c092d90 created [Aug 18 10:34:26] DEBUG[14936] stasis.c: Creating topic. name: cache:617/channel:213184, detail: [Aug 18 10:34:26] DEBUG[14936] stasis.c: Topic 'cache:617/channel:213184': 0x7f0c2c0089c0 created [Aug 18 10:34:26] DEBUG[14966] http.c: match request [ari/playbacks/b52a589b-36b3-45e3-bd52-bf3ffebcf94d] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14964] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14966] http.c: match request [ari/playbacks/b52a589b-36b3-45e3-bd52-bf3ffebcf94d] with handler [phoneprov] len 9 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 From: ;tag=as15514e30 To: ;tag=as11c4e68e Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: Allocating new SIP dialog for 4a4f58147490af446fd9a2593ef6ba48@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14957] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c80099240' [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: Allocating new SIP dialog for 1e1f029e418841157d4eb10724d3434b@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14956] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8415dcf0' [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) RTP allocated port 17618 [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE creating session 0.0.0.0:17618 (17618) [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE create [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE add system candidates [Aug 18 10:34:26] DEBUG[14956] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14956] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE add candidate: 159.65.48.104:17618, 2130706431 [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: Allocating new SIP dialog for 4fa8c7c67315a185220e89483b4b24cd@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14959] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c0e83b0' [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) RTP allocated port 11040 [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE creating session 0.0.0.0:11040 (11040) [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE create [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE add system candidates [Aug 18 10:34:26] DEBUG[14959] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14959] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14956] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14966] http.c: match request [ari/playbacks/b52a589b-36b3-45e3-bd52-bf3ffebcf94d] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) RTP allocated port 11578 [Aug 18 10:34:26] DEBUG[14956] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE add candidate: 10.131.0.10:17618, 2130706431 [Aug 18 10:34:26] DEBUG[14956] rtp_engine.c: RTP instance '0x7f0c8415dcf0' is setup and ready to go [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) ICE stopped [Aug 18 10:34:26] DEBUG[14956] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14956] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14956] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14956] res_rtp_asterisk.c: (0x7f0c8415dcf0) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14956] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14966] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: Allocating new SIP dialog for 6cc7f87b509f9322485af4ea153c327a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14960] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8803fd60' [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) RTP allocated port 19388 [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE creating session 0.0.0.0:19388 (19388) [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE create [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE add system candidates [Aug 18 10:34:26] DEBUG[14960] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14960] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE add candidate: 159.65.48.104:19388, 2130706431 [Aug 18 10:34:26] DEBUG[14960] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14960] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE add candidate: 10.131.0.10:19388, 2130706431 [Aug 18 10:34:26] DEBUG[14960] rtp_engine.c: RTP instance '0x7f0c8803fd60' is setup and ready to go [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) ICE stopped [Aug 18 10:34:26] DEBUG[14960] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14960] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE add candidate: 159.65.48.104:11040, 2130706431 [Aug 18 10:34:26] DEBUG[14959] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14959] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE add candidate: 10.131.0.10:11040, 2130706431 [Aug 18 10:34:26] DEBUG[14959] rtp_engine.c: RTP instance '0x7f0c8c0e83b0' is setup and ready to go [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) ICE stopped [Aug 18 10:34:26] DEBUG[14959] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14959] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14959] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14959] res_rtp_asterisk.c: (0x7f0c8c0e83b0) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14959] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE creating session 0.0.0.0:11578 (11578) [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14959] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14959] chan_sip.c: SIP call-id changed from '4fa8c7c67315a185220e89483b4b24cd@127.0.1.1:5060' to '5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14959] stasis.c: Creating topic. name: channel:213192, detail: [Aug 18 10:34:26] DEBUG[14959] stasis.c: Topic 'channel:213192': 0x7f0c8c0083d0 created [Aug 18 10:34:26] DEBUG[14529] res_rtp_asterisk.c: (0x2c14110) RTCP got report of 76 bytes from 178.62.121.41:15861 [Aug 18 10:34:26] DEBUG[14960] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14960] res_rtp_asterisk.c: (0x7f0c8803fd60) RTCP setup on RTP instance [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as15514e30 [Aug 18 10:34:26] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE create [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Finding handler for playbacks/b52a589b-36b3-45e3-bd52-bf3ffebcf94d [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Finding handler for bridges [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Finding handler for bridges [Aug 18 10:34:26] DEBUG[14745] res_rtp_asterisk.c: (0x7f0c880b67b0) RTP audio difference is 928, ms is 78 [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14939] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE add system candidates [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14959] stasis.c: Creating topic. name: cache:618/channel:213192, detail: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as11c4e68e [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:26] DEBUG[14956] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[13682] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14957] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Finding handler for playbacks [Aug 18 10:34:26] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" [Aug 18 10:34:26] VERBOSE[14960] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14956] chan_sip.c: SIP call-id changed from '1e1f029e418841157d4eb10724d3434b@127.0.1.1:5060' to '6ba2731c352a981429345b965d229ce4@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14956] stasis.c: Creating topic. name: channel:213190, detail: [Aug 18 10:34:26] DEBUG[14956] stasis.c: Topic 'channel:213190': 0x7f0c8407ddd0 created [Aug 18 10:34:26] DEBUG[14956] stasis.c: Creating topic. name: cache:619/channel:213190, detail: [Aug 18 10:34:26] DEBUG[14956] stasis.c: Topic 'cache:619/channel:213190': 0x7f0c8409a280 created [Aug 18 10:34:26] DEBUG[14959] stasis.c: Topic 'cache:618/channel:213192': 0x7f0c8c03ff70 created [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 (Checking To) --From tag as15514e30 --To-tag as11c4e68e [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (1) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (1) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: Allocating new SIP dialog for 5a2604157aa565f003ddd1ac2bad0b46@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:26] DEBUG[14957] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE add candidate: 159.65.48.104:11578, 2130706431 [Aug 18 10:34:26] DEBUG[14957] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14957] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE add candidate: 10.131.0.10:11578, 2130706431 [Aug 18 10:34:26] DEBUG[14957] rtp_engine.c: RTP instance '0x7f0c80099240' is setup and ready to go [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) ICE stopped [Aug 18 10:34:26] DEBUG[14957] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14957] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:26] DEBUG[14950] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c08df30' [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) RTP allocated port 17348 [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE creating session 0.0.0.0:17348 (17348) [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE create [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE add system candidates [Aug 18 10:34:26] DEBUG[14950] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14950] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE add candidate: 159.65.48.104:17348, 2130706431 [Aug 18 10:34:26] DEBUG[14950] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14950] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE add candidate: 10.131.0.10:17348, 2130706431 [Aug 18 10:34:26] DEBUG[14950] rtp_engine.c: RTP instance '0x7f0c7c08df30' is setup and ready to go [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) ICE stopped [Aug 18 10:34:26] DEBUG[14950] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14950] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14950] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14950] res_rtp_asterisk.c: (0x7f0c7c08df30) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14950] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14957] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:26] DEBUG[14950] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Finding handler for b52a589b-36b3-45e3-bd52-bf3ffebcf94d [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:26] DEBUG[14966] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14964] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:26] DEBUG[14957] res_rtp_asterisk.c: (0x7f0c80099240) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14957] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14966] res_ari.c: No explicit handler found for b52a589b-36b3-45e3-bd52-bf3ffebcf94d. Using wildcard playbackId. [Aug 18 10:34:26] DEBUG[14964] stasis.c: Creating topic. name: bridge:99977815-16b9-491f-b96b-6fba43c3cc45, detail: [Aug 18 10:34:26] DEBUG[14964] stasis.c: Topic 'bridge:99977815-16b9-491f-b96b-6fba43c3cc45': 0x7f0ca811b1c0 created [Aug 18 10:34:26] DEBUG[14964] stasis.c: Creating topic. name: cache:620/bridge:99977815-16b9-491f-b96b-6fba43c3cc45, detail: [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14964] stasis.c: Topic 'cache:620/bridge:99977815-16b9-491f-b96b-6fba43c3cc45': 0x7f0ca810c090 created [Aug 18 10:34:26] DEBUG[14960] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14950] chan_sip.c: SIP call-id changed from '5a2604157aa565f003ddd1ac2bad0b46@127.0.1.1:5060' to '40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14960] chan_sip.c: SIP call-id changed from '6cc7f87b509f9322485af4ea153c327a@127.0.1.1:5060' to '21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: Allocating new SIP dialog for 0d6b7483740e684758d67eb441471200@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14962] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c940d7f80' [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: Allocating new SIP dialog for 5280370a7121404a2c3c5d143d05e2a8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14964] bridge_native_rtp.c: Bridge '99977815-16b9-491f-b96b-6fba43c3cc45' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[14964] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[14964] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[14964] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:26] DEBUG[14964] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[14964] bridge.c: Bridge 99977815-16b9-491f-b96b-6fba43c3cc45: calling simple_bridge technology constructor [Aug 18 10:34:26] DEBUG[14964] bridge.c: Bridge 99977815-16b9-491f-b96b-6fba43c3cc45: calling simple_bridge technology start [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (1) BYE - 8 [Aug 18 10:34:26] DEBUG[14957] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14957] chan_sip.c: SIP call-id changed from '4a4f58147490af446fd9a2593ef6ba48@127.0.1.1:5060' to '7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) RTP allocated port 13644 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:26] DEBUG[14960] stasis.c: Creating topic. name: channel:213189, detail: [Aug 18 10:34:26] DEBUG[14960] stasis.c: Topic 'channel:213189': 0x7f0c8807d1a0 created [Aug 18 10:34:26] DEBUG[14960] stasis.c: Creating topic. name: cache:621/channel:213189, detail: [Aug 18 10:34:26] DEBUG[14960] stasis.c: Topic 'cache:621/channel:213189': 0x7f0c8803aea0 created [Aug 18 10:34:26] DEBUG[14943] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c40006350' [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) RTP allocated port 14186 [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE creating session 0.0.0.0:14186 (14186) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[13682] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE create [Aug 18 10:34:26] DEBUG[14957] stasis.c: Creating topic. name: channel:213188, detail: [Aug 18 10:34:26] DEBUG[14950] stasis.c: Creating topic. name: channel:213186, detail: [Aug 18 10:34:26] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:26] DEBUG[14968] http.c: HTTP opening session. Top level [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14966] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:26] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14966] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE add system candidates [Aug 18 10:34:26] DEBUG[14943] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE creating session 0.0.0.0:13644 (13644) [Aug 18 10:34:26] DEBUG[13682] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:26] DEBUG[13682] channel.c: Channel Announcer/ARI-0000001b;1 setting write format path: slin -> slin [Aug 18 10:34:26] NOTICE[13682] res_stasis_playback.c: 1629282838.161: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:26] DEBUG[13682] channel.c: Channel 0x7f0c70070730 'Announcer/ARI-0000001b;1' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:26] DEBUG[14943] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14968] http.c: HTTP Request URI is /ari/channels/robot_212995 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14964] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:26] DEBUG[14964] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: Allocating new SIP dialog for 654cf9912ed374637a31d768150fb77e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14955] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7803d4a0' [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) RTP allocated port 12262 [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE creating session 0.0.0.0:12262 (12262) [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE create [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE add system candidates [Aug 18 10:34:26] DEBUG[14955] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14955] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE add candidate: 159.65.48.104:12262, 2130706431 [Aug 18 10:34:26] DEBUG[14955] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14955] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE add candidate: 10.131.0.10:12262, 2130706431 [Aug 18 10:34:26] DEBUG[14955] rtp_engine.c: RTP instance '0x7f0c7803d4a0' is setup and ready to go [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) ICE stopped [Aug 18 10:34:26] DEBUG[14955] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14955] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (2) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:26] DEBUG[14950] stasis.c: Topic 'channel:213186': 0x7f0c7c053f30 created [Aug 18 10:34:26] DEBUG[14955] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14969] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14957] stasis.c: Topic 'channel:213188': 0x7f0c800a9650 created [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14969] http.c: HTTP Request URI is /ari/channels/212984/snoop?app=calls_0&spy=in [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE add candidate: 159.65.48.104:14186, 2130706431 [Aug 18 10:34:26] DEBUG[14968] http.c: match request [ari/channels/robot_212995] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE create [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:26] DEBUG[14968] http.c: match request [ari/channels/robot_212995] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14955] res_rtp_asterisk.c: (0x7f0c7803d4a0) RTCP setup on RTP instance [Aug 18 10:34:26] DEBUG[14738] res_rtp_asterisk.c: (0x7f0c900475d0) RTP audio difference is 736, ms is 66 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14969] http.c: match request [ari/channels/212984/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14950] stasis.c: Creating topic. name: cache:622/channel:213186, detail: [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE add system candidates [Aug 18 10:34:26] DEBUG[14968] http.c: match request [ari/channels/robot_212995] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14968] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14950] stasis.c: Topic 'cache:622/channel:213186': 0x7f0c7c094c80 created [Aug 18 10:34:26] DEBUG[14969] http.c: match request [ari/channels/212984/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14969] http.c: match request [ari/channels/212984/snoop] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14957] stasis.c: Creating topic. name: cache:623/channel:213188, detail: [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14962] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14969] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[13354] channel.c: Channel 0x7f0c1c012020 'Recorder/ARI-0000000c;1' destroying [Aug 18 10:34:26] DEBUG[14727] channel.c: Channel 0x7f0c3c1372e0 'Announcer/ARI-0000003a;1' destroying [Aug 18 10:34:26] DEBUG[14727] stasis.c: Destroying topic. name: cache:422/channel:1629282854.366, detail: [Aug 18 10:34:26] DEBUG[14727] stasis.c: Topic 'cache:422/channel:1629282854.366': 0x7f0c3c13aef0 destroyed [Aug 18 10:34:26] DEBUG[14727] stasis.c: Destroying topic. name: channel:1629282854.366, detail: [Aug 18 10:34:26] DEBUG[14727] stasis.c: Topic 'channel:1629282854.366': 0x7f0c3c13a4c0 destroyed [Aug 18 10:34:26] DEBUG[14722] bridge_channel.c: Setting 0x7f0c3c136e60(Announcer/ARI-0000003a;2) state from:0 to:1 [Aug 18 10:34:26] DEBUG[14722] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pulling 0x7f0c3c136e60(Announcer/ARI-0000003a;2) [Aug 18 10:34:26] VERBOSE[14722] bridge_channel.c: Channel Announcer/ARI-0000003a;2 left 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:26] DEBUG[14722] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c3c136e60(Announcer/ARI-0000003a;2) is leaving simple_bridge technology [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Finding handler for channels/robot_212995 [Aug 18 10:34:26] DEBUG[14962] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50118 - state 0 (Unknown) [Aug 18 10:34:26] VERBOSE[14955] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14943] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14722] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14955] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14955] chan_sip.c: SIP call-id changed from '654cf9912ed374637a31d768150fb77e@127.0.1.1:5060' to '5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14955] stasis.c: Creating topic. name: channel:213187, detail: [Aug 18 10:34:26] DEBUG[14955] stasis.c: Topic 'channel:213187': 0x7f0c7803b0e0 created [Aug 18 10:34:26] DEBUG[14955] stasis.c: Creating topic. name: cache:624/channel:213187, detail: [Aug 18 10:34:26] DEBUG[14955] stasis.c: Topic 'cache:624/channel:213187': 0x7f0c7807d860 created [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14722] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Finding handler for channels/212984/snoop [Aug 18 10:34:26] DEBUG[13354] stasis.c: Destroying topic. name: cache:116/channel:1629282833.97, detail: [Aug 18 10:34:26] DEBUG[13354] stasis.c: Topic 'cache:116/channel:1629282833.97': 0x7f0c1c017750 destroyed [Aug 18 10:34:26] DEBUG[13354] stasis.c: Destroying topic. name: channel:1629282833.97, detail: [Aug 18 10:34:26] DEBUG[13354] stasis.c: Topic 'channel:1629282833.97': 0x7f0c1c01e760 destroyed [Aug 18 10:34:26] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50118, detail: [Aug 18 10:34:26] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50118': 0x7f0c84108fd0 created [Aug 18 10:34:26] DEBUG[13626] channel.c: Channel 0x7f0c80047da0 'Recorder/ARI-0000001a;2' destroying [Aug 18 10:34:26] DEBUG[13361] channel.c: Channel 0x7f0c40010a50 'SIP/zvonobot-00000008' destroying [Aug 18 10:34:26] DEBUG[14722] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14957] stasis.c: Topic 'cache:623/channel:213188': 0x7f0c80047c20 created [Aug 18 10:34:26] DEBUG[14722] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:26] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50118' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Aug 18 10:34:26] DEBUG[14722] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14722] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:26] VERBOSE[13638] app.c: User hung up [Aug 18 10:34:26] DEBUG[13638] res_stasis_recording.c: 1629282837.155: Recording complete [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Recorder - ARI [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 From: ;tag=as22d5765f To: ;tag=as0550790a Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[14943] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE add candidate: 159.65.48.104:13644, 2130706431 [Aug 18 10:34:26] DEBUG[13638] channel.c: Channel 0x7f0c800507f0 'Recorder/ARI-0000001a;1' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282866.545, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.545': 0x7f0c301438a0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: cache:625/channel:1629282866.545, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:625/channel:1629282866.545': 0x7f0c300799b0 created [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE add candidate: 10.131.0.10:14186, 2130706431 [Aug 18 10:34:26] DEBUG[14943] rtp_engine.c: RTP instance '0x7f0c40006350' is setup and ready to go [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) ICE stopped [Aug 18 10:34:26] DEBUG[14943] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14943] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14943] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14943] res_rtp_asterisk.c: (0x7f0c40006350) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14943] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14943] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14943] chan_sip.c: SIP call-id changed from '5280370a7121404a2c3c5d143d05e2a8@127.0.1.1:5060' to '21405c0201d61407119338763dc16673@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14943] stasis.c: Creating topic. name: channel:213185, detail: [Aug 18 10:34:26] DEBUG[14943] stasis.c: Topic 'channel:213185': 0x7f0c400563c0 created [Aug 18 10:34:26] DEBUG[14943] stasis.c: Creating topic. name: cache:626/channel:213185, detail: [Aug 18 10:34:26] DEBUG[14943] stasis.c: Topic 'cache:626/channel:213185': 0x7f0c40006870 created [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel '212972': is 0 interested in calls_0 [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel '212972' unsubscribed from calls_0 [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Finding handler for robot_212995 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22d5765f [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking channels create: Didn't match robot_212995 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0550790a [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: cache:625/channel:1629282866.545, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:625/channel:1629282866.545': 0x7f0c300799b0 destroyed [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282866.545, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.545': 0x7f0c301438a0 destroyed [Aug 18 10:34:26] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000008', '', 'Stasis', 'calls_0', 37, 26, 'ANSWERED', 3, '', '212972', '')] [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14722] channel.c: Channel 0x7f0c3c03b3f0 'Announcer/ARI-0000003a;2' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 (Checking To) --From tag as22d5765f --To-tag as0550790a [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (2) BYE - 8 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (4) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[13626] stasis.c: Destroying topic. name: cache:188/channel:1629282837.158, detail: [Aug 18 10:34:26] DEBUG[13361] channel.c: Channel 0x7f0c74028ad0 'Snoop/212972-00000006' destroying [Aug 18 10:34:26] DEBUG[13626] stasis.c: Topic 'cache:188/channel:1629282837.158': 0x7f0c80052830 destroyed [Aug 18 10:34:26] DEBUG[14962] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[13626] stasis.c: Destroying topic. name: channel:1629282837.158, detail: [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14962] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE add candidate: 10.131.0.10:13644, 2130706431 [Aug 18 10:34:26] DEBUG[14962] rtp_engine.c: RTP instance '0x7f0c940d7f80' is setup and ready to go [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) ICE stopped [Aug 18 10:34:26] DEBUG[14962] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[14962] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[13626] stasis.c: Topic 'channel:1629282837.158': 0x7f0c800526e0 destroyed [Aug 18 10:34:26] WARNING[14665] app.c: No audio available on Recorder/ARI-00000035;1?? [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] VERBOSE[14665] app.c: User hung up [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[20620] stasis.c: Destroying topic. name: cache:15/channel:212972, detail: [Aug 18 10:34:26] DEBUG[20620] stasis.c: Topic 'cache:15/channel:212972': 0x7f0c400123a0 destroyed [Aug 18 10:34:26] DEBUG[20620] stasis.c: Destroying topic. name: channel:212972, detail: [Aug 18 10:34:26] DEBUG[20620] stasis.c: Topic 'channel:212972': 0x7f0c40076ff0 destroyed [Aug 18 10:34:26] DEBUG[14665] res_stasis_recording.c: 1629282853.345: Recording complete [Aug 18 10:34:26] DEBUG[14962] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14968] res_ari.c: Checking channels externalMedia: Didn't match robot_212995 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Finding handler for 212984 [Aug 18 10:34:26] DEBUG[14665] channel.c: Channel 0x7f0c3c12ca30 'Recorder/ARI-00000035;1' hanging up. Refs: 2 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channels create: Didn't match 212984 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channels externalMedia: Didn't match 212984 [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282866.547, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.547': 0x7f0c300799b0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: cache:627/channel:1629282866.547, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:627/channel:1629282866.547': 0x7f0c300925e0 created [Aug 18 10:34:26] DEBUG[14969] res_ari.c: No explicit handler found for 212984. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14887] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14968] res_ari.c: No explicit handler found for robot_212995. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[14962] res_rtp_asterisk.c: (0x7f0c940d7f80) RTCP setup on RTP instance [Aug 18 10:34:26] DEBUG[13361] stasis.c: Destroying topic. name: cache:123/channel:1629282833.102, detail: [Aug 18 10:34:26] DEBUG[13361] stasis.c: Topic 'cache:123/channel:1629282833.102': 0x7f0c74022cc0 destroyed [Aug 18 10:34:26] DEBUG[13361] stasis.c: Destroying topic. name: channel:1629282833.102, detail: [Aug 18 10:34:26] DEBUG[13361] stasis.c: Topic 'channel:1629282833.102': 0x7f0c740229f0 destroyed [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Finding handler for snoop [Aug 18 10:34:26] VERBOSE[14962] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14962] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: cache:627/channel:1629282866.547, detail: [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:26] DEBUG[14416] channel.c: Channel 0x7f0c9c0ab760 'Announcer/ARI-0000002f;2' destroying [Aug 18 10:34:26] DEBUG[14787] channel.c: Channel 0x7f0c3008a100 'SIP/zvonobot-000000d0' allocated [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14797] channel.c: Channel 0x7f0c84148b70 'Announcer/ARI-0000004a;1' allocated [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:26] DEBUG[14804] channel.c: Channel 0x7f0c9c09c340 'Recorder/ARI-0000004b;1' allocated [Aug 18 10:34:26] DEBUG[14804] stasis.c: Creating topic. name: channel:1629282866.549, detail: [Aug 18 10:34:26] DEBUG[14804] stasis.c: Topic 'channel:1629282866.549': 0x7f0c9c05ea90 created [Aug 18 10:34:26] DEBUG[14804] stasis.c: Creating topic. name: cache:628/channel:1629282866.549, detail: [Aug 18 10:34:26] DEBUG[14804] stasis.c: Topic 'cache:628/channel:1629282866.549': 0x7f0c9c047ca0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:627/channel:1629282866.547': 0x7f0c300925e0 destroyed [Aug 18 10:34:26] DEBUG[14797] stasis.c: Creating topic. name: channel:1629282866.548, detail: [Aug 18 10:34:26] DEBUG[14797] stasis.c: Topic 'channel:1629282866.548': 0x7f0c840780a0 created [Aug 18 10:34:26] DEBUG[14797] stasis.c: Creating topic. name: cache:629/channel:1629282866.548, detail: [Aug 18 10:34:26] DEBUG[14797] stasis.c: Topic 'cache:629/channel:1629282866.548': 0x7f0c84049bf0 created [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:26] DEBUG[14787] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:26] DEBUG[14416] stasis.c: Destroying topic. name: cache:339/channel:1629282849.291, detail: [Aug 18 10:34:26] DEBUG[14416] stasis.c: Topic 'cache:339/channel:1629282849.291': 0x7f0c9c0082e0 destroyed [Aug 18 10:34:26] DEBUG[14416] stasis.c: Destroying topic. name: channel:1629282849.291, detail: [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:26] DEBUG[14416] stasis.c: Topic 'channel:1629282849.291': 0x7f0c9c035ab0 destroyed [Aug 18 10:34:26] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282866.547, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.547': 0x7f0c300799b0 destroyed [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] WARNING[13678] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-0000001c;1 [Aug 18 10:34:26] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'Snoop/212972-00000006', 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0', 'Stasis', 'calls_0', 29, 29, 'ANSWERED', 3, '', '1629282833.102', '')] [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16;received=159.65.48.104 From: ;tag=as1885cc1f To: ;tag=as5ecc8581 Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ed5b4b0" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[14962] chan_sip.c: SIP call-id changed from '0d6b7483740e684758d67eb441471200@127.0.1.1:5060' to '5676735873320902534290d27970f7c7@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14962] stasis.c: Creating topic. name: channel:213193, detail: [Aug 18 10:34:26] DEBUG[14962] stasis.c: Topic 'channel:213193': 0x7f0c94047c20 created [Aug 18 10:34:26] DEBUG[14962] stasis.c: Creating topic. name: cache:630/channel:213193, detail: [Aug 18 10:34:26] DEBUG[14962] stasis.c: Topic 'cache:630/channel:213193': 0x7f0c940f01f0 created [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[14788] channel.c: Channel 0x7f0c3c176140 'SIP/zvonobot-000000d1' allocated [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:26] DEBUG[14788] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:26] DEBUG[13377] res_rtp_asterisk.c: (0x7f0c840529d0) DTLS stop [Aug 18 10:34:26] DEBUG[13377] res_rtp_asterisk.c: (0x7f0c840529d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[13377] res_rtp_asterisk.c: (0x7f0c840529d0) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[13377] res_rtp_asterisk.c: (0x7f0c840529d0) ICE stopped [Aug 18 10:34:26] DEBUG[13377] rtp_engine.c: Destroyed RTP instance '0x7f0c840529d0' [Aug 18 10:34:26] DEBUG[13377] channel.c: Channel 0x7f0c84060000 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0' destroying [Aug 18 10:34:26] DEBUG[14788] res_stasis.c: calls_0: Subscribing to 213172 [Aug 18 10:34:26] DEBUG[14788] stasis/app.c: Channel '213172' is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:26] DEBUG[14319] channel.c: Channel 0x7f0c180f2f90 'Recorder/ARI-00000030;2' destroying [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[14788] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:26] DEBUG[14788] http.c: HTTP closing session. Top level [Aug 18 10:34:26] DEBUG[14319] stasis.c: Destroying topic. name: cache:340/channel:1629282849.293, detail: [Aug 18 10:34:26] DEBUG[14319] stasis.c: Topic 'cache:340/channel:1629282849.293': 0x7f0c180d8b10 destroyed [Aug 18 10:34:26] DEBUG[14319] stasis.c: Destroying topic. name: channel:1629282849.293, detail: [Aug 18 10:34:26] DEBUG[14319] stasis.c: Topic 'channel:1629282849.293': 0x7f0c180d88e0 destroyed [Aug 18 10:34:26] DEBUG[14400] chan_sip.c: Hangup call SIP/zvonobot-00000094, SIP callid 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14400] res_rtp_asterisk.c: (0x7f0c940c4d70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[14400] res_rtp_asterisk.c: (0x7f0c940c4d70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[14400] channel.c: Channel 0x7f0c940d16b0 'SIP/zvonobot-00000094' destroying [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Outgoing Call for 79821116868 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel 'robot_212972': is 0 interested in calls_0 [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel 'robot_212972' unsubscribed from calls_0 [Aug 18 10:34:26] DEBUG[20620] stasis.c: Destroying topic. name: cache:129/channel:robot_212972, detail: [Aug 18 10:34:26] DEBUG[20620] stasis.c: Topic 'cache:129/channel:robot_212972': 0x7f0c84062c40 destroyed [Aug 18 10:34:26] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212972, detail: [Aug 18 10:34:26] DEBUG[20620] stasis.c: Topic 'channel:robot_212972': 0x7f0c8405fd40 destroyed [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1885cc1f [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5ecc8581 [Aug 18 10:34:26] DEBUG[14787] res_stasis.c: calls_0: Subscribing to 213171 [Aug 18 10:34:26] DEBUG[14787] stasis/app.c: Channel '213171' is 1 interested in calls_0 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ed5b4b0" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 (Checking To) --From tag as1885cc1f --To-tag as5ecc8581 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '780fccc405c242d348e2e247384adc25@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (5) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116909@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7b11e1d4 Max-Forwards: 70 From: ;tag=as3097acc5 To: Contact: Call-ID: 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 693565206 693565206 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10524 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (4) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (4) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel '213110': is 0 interested in calls_0 [Aug 18 10:34:26] DEBUG[20620] stasis/app.c: channel '213110' unsubscribed from calls_0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (5) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Outgoing Call for 79821116869 [Aug 18 10:34:26] DEBUG[14787] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282866.551, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.551': 0x7f0c300311f0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: cache:631/channel:1629282866.551, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:631/channel:1629282866.551': 0x7f0c300925e0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: cache:631/channel:1629282866.551, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:631/channel:1629282866.551': 0x7f0c300925e0 destroyed [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282866.551, detail: [Aug 18 10:34:26] DEBUG[14400] stasis.c: Destroying topic. name: cache:358/channel:213110, detail: [Aug 18 10:34:26] DEBUG[14400] stasis.c: Topic 'cache:358/channel:213110': 0x7f0c940d3e70 destroyed [Aug 18 10:34:26] DEBUG[14400] stasis.c: Destroying topic. name: channel:213110, detail: [Aug 18 10:34:26] DEBUG[14400] stasis.c: Topic 'channel:213110': 0x7f0c940d3430 destroyed [Aug 18 10:34:26] DEBUG[14972] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: Allocating new SIP dialog for 071a4d165651f7a10a51830f3fefdb11@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:26] DEBUG[14965] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c058920' [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) RTP allocated port 18212 [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE creating session 0.0.0.0:18212 (18212) [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE create [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE add system candidates [Aug 18 10:34:26] DEBUG[14972] http.c: HTTP Request URI is /ari/channels/213110 [Aug 18 10:34:26] DEBUG[14972] http.c: match request [ari/channels/213110] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14972] http.c: match request [ari/channels/213110] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.551': 0x7f0c300311f0 destroyed [Aug 18 10:34:26] DEBUG[14972] http.c: match request [ari/channels/213110] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14965] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:26] DEBUG[14972] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14965] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:26] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50283-0x7f0c840529d0', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212972', '')] [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE add candidate: 159.65.48.104:18212, 2130706431 [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Finding handler for channels/213110 [Aug 18 10:34:26] DEBUG[14965] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:26] DEBUG[14965] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Finding handler for channels [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE add candidate: 10.131.0.10:18212, 2130706431 [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:26] DEBUG[14965] rtp_engine.c: RTP instance '0x7f0c9c058920' is setup and ready to go [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Finding handler for 213110 [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) ICE stopped [Aug 18 10:34:26] DEBUG[14884] app.c: One waitfor failed, trying another [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking channels create: Didn't match 213110 [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14972] res_ari.c: Checking channels externalMedia: Didn't match 213110 [Aug 18 10:34:26] DEBUG[14972] res_ari.c: No explicit handler found for 213110. Using wildcard channelId. [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14787] http.c: HTTP closing session. Top level [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK435e0a94;received=159.65.48.104 From: ;tag=as6658275e To: ;tag=as7c4646b4 Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 262 v=0 o=root 72351813 72351813 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12564 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK435e0a94;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6658275e [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7c4646b4 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 1 [ 45]: o=root 72351813 72351813 IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12564 RTP/AVP 0 8 101 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 (Checking To) --From tag as6658275e --To-tag as7c4646b4 [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:26] DEBUG[14969] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '0a35965008fea95b4665220a212af999@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282866.552, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.552': 0x7f0c300925e0 created [Aug 18 10:34:26] DEBUG[20545] stasis.c: Creating topic. name: cache:632/channel:1629282866.552, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:632/channel:1629282866.552': 0x7f0c300311f0 created [Aug 18 10:34:26] DEBUG[14965] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: cache:632/channel:1629282866.552, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'cache:632/channel:1629282866.552': 0x7f0c300311f0 destroyed [Aug 18 10:34:26] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282866.552, detail: [Aug 18 10:34:26] DEBUG[20545] stasis.c: Topic 'channel:1629282866.552': 0x7f0c300925e0 destroyed [Aug 18 10:34:26] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:13', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000094', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213110', '')] [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Got SDP version 72351813 and unique parts [root 72351813 IN IP4 178.62.121.41] [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 72351813 72351813 IN IP4 178.62.121.41... OK. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:26] DEBUG[14965] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:26] VERBOSE[14971] chan_sip.c: Audio is at 17014 [Aug 18 10:34:26] VERBOSE[14971] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:26] VERBOSE[14971] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:26] VERBOSE[14971] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Initializing initreq for method INVITE - callid 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116869@178.62.121.41 SIP/2.0 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 3 [ 52]: From: ;tag=as73a421e4 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 6 [ 60]: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:26 GMT [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] VERBOSE[14971] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28 [Aug 18 10:34:26] DEBUG[14971] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c008d30) ICE set role failed; no ice instance [Aug 18 10:34:26] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c008d30) RTCP setting address on RTP instance [Aug 18 10:34:26] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c2c05ce80 -- Strict RTP learning after remote address set to: 178.62.121.41:12564 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:12564 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb003fbf8) from 0x7f0c147e2330 to 0x7f0c2c008f08 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb00795f8) from 0x7f0c147e2330 to 0x7f0c2c008f08 [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb0089a68) from 0x7f0c147e2330 to 0x7f0c2c008f08 [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c008d30) RTCP ignoring duplicate property [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:26] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003e setting read format path: alaw -> alaw [Aug 18 10:34:26] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003e setting write format path: alaw -> alaw [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c008d30) DTLS - ast_rtp_activate rtp=0x7f0c2c05ce80 - setup and perform DTLS' [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c05ce80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c05ce80) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:26] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Strict routing enforced for session 0a35965008fea95b4665220a212af999@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:26] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:26] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117015@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7fa75d75 Max-Forwards: 70 From: ;tag=as6658275e To: ;tag=as7c4646b4 Contact: Call-ID: 0a35965008fea95b4665220a212af999@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[14965] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:26] DEBUG[14965] res_rtp_asterisk.c: (0x7f0c9c058920) RTCP setup on RTP instance [Aug 18 10:34:26] VERBOSE[14965] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:26] DEBUG[14965] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:26] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] VERBOSE[14970] chan_sip.c: Audio is at 14666 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14965] chan_sip.c: SIP call-id changed from '071a4d165651f7a10a51830f3fefdb11@127.0.1.1:5060' to '64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060' [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] VERBOSE[14970] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Destroying SIP dialog 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:26] VERBOSE[13415] dial.c: SIP/zvonobot-0000003e answered [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940c4d70) DTLS stop [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940c4d70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] VERBOSE[13415] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003e [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:26] VERBOSE[14970] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:26] VERBOSE[14970] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Initializing initreq for method INVITE - callid 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940c4d70) DTLS srtp - stopped timeout timer' [Aug 18 10:34:26] DEBUG[14965] stasis.c: Creating topic. name: channel:213191, detail: [Aug 18 10:34:26] DEBUG[14965] stasis.c: Topic 'channel:213191': 0x7f0c9c0e63f0 created [Aug 18 10:34:26] VERBOSE[14971] dial.c: Called zvonobot/79821116869 [Aug 18 10:34:26] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c940c4d70) ICE RTP transport deallocating [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116868@178.62.121.41 SIP/2.0 [Aug 18 10:34:26] DEBUG[13415] stasis/app.c: Channel '213025' is 2 interested in calls_0 [Aug 18 10:34:26] DEBUG[14965] stasis.c: Creating topic. name: cache:633/channel:213191, detail: [Aug 18 10:34:26] VERBOSE[13415] res_rtp_asterisk.c: 0x7f0c2c05ce80 -- Strict RTP switching to RTP target address 178.62.121.41:12564 as source [Aug 18 10:34:26] DEBUG[13415] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:26] DEBUG[13415] channel.c: Channel SIP/zvonobot-0000003e setting read format path: ulaw -> alaw [Aug 18 10:34:26] DEBUG[13415] channel.c: Channel SIP/zvonobot-0000003e setting write format path: alaw -> ulaw [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:26] DEBUG[14965] stasis.c: Topic 'cache:633/channel:213191': 0x7f0c9c068a60 created [Aug 18 10:34:26] DEBUG[14974] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 3 [ 52]: From: ;tag=as406d4539 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 6 [ 60]: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:26 GMT [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:26] VERBOSE[14970] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c940c4d70' [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Session timer started: 7 - 0a35965008fea95b4665220a212af999@159.65.48.104:5060 1768000ms [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 From: ;tag=as7dd13c21 To: ;tag=as19c9362c Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7dd13c21 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as19c9362c [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 (Checking To) --From tag as7dd13c21 --To-tag as19c9362c [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #78 [Aug 18 10:34:26] DEBUG[14970] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14974] http.c: HTTP Request URI is /ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:26] DEBUG[14974] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [httpstatus] len 10 [Aug 18 10:34:26] DEBUG[14974] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [phoneprov] len 9 [Aug 18 10:34:26] DEBUG[14974] http.c: match request [ari/bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play] with handler [ari] len 3 [Aug 18 10:34:26] DEBUG[14974] http.c: Match made with [ari] [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Finding handler for bridges/e8b160c4-f3ae-46ad-bda0-ffa245693ffa/play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Finding handler for bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Finding handler for e8b160c4-f3ae-46ad-bda0-ffa245693ffa [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:26] DEBUG[14974] res_ari.c: No explicit handler found for e8b160c4-f3ae-46ad-bda0-ffa245693ffa. Using wildcard bridgeId. [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Finding handler for play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:26] DEBUG[14974] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:26] VERBOSE[14970] dial.c: Called zvonobot/79821116868 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116908@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af Max-Forwards: 70 From: ;tag=as7dd13c21 To: Contact: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[14975] http.c: HTTP opening session. Top level [Aug 18 10:34:26] DEBUG[14931] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (3) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:26] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456;received=159.65.48.104 From: ;tag=as307f6396 To: ;tag=as3a0db143 Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37ef5652" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK735e5456;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as307f6396 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3a0db143 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37ef5652" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 (Checking To) --From tag as307f6396 --To-tag as3a0db143 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4aae0db47b6087472605fe692ef6a9b3@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (3) BYE - 8 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc;received=159.65.48.104 From: ;tag=as20932b4d To: ;tag=as2141137d Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d23c7e2" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as20932b4d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2141137d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[14745] res_rtp_asterisk.c: (0x7f0c880b67b0) RTP audio difference is 784, ms is 69 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d23c7e2" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 (Checking To) --From tag as20932b4d --To-tag as2141137d [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116912@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04c630dc Max-Forwards: 70 From: ;tag=as20932b4d To: Contact: Call-ID: 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (1) INVITE - 5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 From: ;tag=as671c682b To: ;tag=as53b926c5 Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as671c682b [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as53b926c5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 (Checking To) --From tag as671c682b --To-tag as53b926c5 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '2749fa7d41ec862f1556002a63546011@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116916@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b Max-Forwards: 70 From: ;tag=as671c682b To: Contact: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] DEBUG[14975] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16;received=159.65.48.104 From: ;tag=as1885cc1f To: ;tag=as5ecc8581 Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ed5b4b0" Content-Length: 0 <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16;received=159.65.48.104 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1885cc1f [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5ecc8581 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ed5b4b0" [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: = Looking for Call ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 (Checking To) --From tag as1885cc1f --To-tag as5ecc8581 [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Stopping retransmission on '780fccc405c242d348e2e247384adc25@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116926@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0d5c0b16 Max-Forwards: 70 From: ;tag=as1885cc1f To: Contact: Call-ID: 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:26] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 From: ;tag=as54e004b1 To: ;tag=as58786f9e Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1451248754 1451248754 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17184 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:26] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:26] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:26] DEBUG[14975] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[14738] res_rtp_asterisk.c: (0x7f0c900475d0) RTP audio difference is 768, ms is 68 [Aug 18 10:34:26] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54e004b1 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as58786f9e [Aug 18 10:34:27] DEBUG[14669] channel.c: Channel 0x7f0c8c100940 'Snoop/213006-00000016' allocated [Aug 18 10:34:27] DEBUG[14669] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[14667] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14991] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[14991] http.c: HTTP Request URI is /ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/play?media=sound%3Asilence%2F2 [Aug 18 10:34:27] DEBUG[14991] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/play] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14991] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/play] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14991] http.c: match request [ari/bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/play] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14991] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Finding handler for bridges/94ef4fc9-246b-4999-9567-b41f8ba44681/play [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[14669] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:27] DEBUG[14975] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14667] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14975] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14975] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Finding handler for 94ef4fc9-246b-4999-9567-b41f8ba44681 [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[14991] res_ari.c: No explicit handler found for 94ef4fc9-246b-4999-9567-b41f8ba44681. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[14998] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Finding handler for play [Aug 18 10:34:27] DEBUG[14975] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[14995] http.c: HTTP Request URI is /ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play?media=sound%3Asilence%2F2 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1451248754 1451248754 IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:27] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 896, ms is 76 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:27] DEBUG[14995] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14998] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212996&app=calls_0&format=slin16&external_host=127.0.0.1%3A50312 [Aug 18 10:34:27] DEBUG[14975] stasis.c: Creating topic. name: bridge:3e9f0826-47ef-4258-a05a-53af5ce8577c, detail: [Aug 18 10:34:27] DEBUG[14995] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14995] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14995] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Finding handler for bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Finding handler for fa4500bd-5aa0-4ef0-bd71-2993d47cd759 [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[14995] res_ari.c: No explicit handler found for fa4500bd-5aa0-4ef0-bd71-2993d47cd759. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Finding handler for play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:27] DEBUG[14995] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:27] DEBUG[14995] stasis.c: Creating topic. name: channel:1629282867.554, detail: [Aug 18 10:34:27] DEBUG[14995] stasis.c: Topic 'channel:1629282867.554': 0x7f0c180c9550 created [Aug 18 10:34:27] DEBUG[14995] stasis.c: Creating topic. name: cache:634/channel:1629282867.554, detail: [Aug 18 10:34:27] DEBUG[14995] stasis.c: Topic 'cache:634/channel:1629282867.554': 0x7f0c18015e30 created [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:27] DEBUG[14991] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:27] DEBUG[14991] stasis.c: Creating topic. name: channel:1629282867.555, detail: [Aug 18 10:34:27] DEBUG[14991] stasis.c: Topic 'channel:1629282867.555': 0x7f0c1c012430 created [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[14991] stasis.c: Creating topic. name: cache:635/channel:1629282867.555, detail: [Aug 18 10:34:27] DEBUG[14991] stasis.c: Topic 'cache:635/channel:1629282867.555': 0x7f0c1c012d60 created [Aug 18 10:34:27] DEBUG[14394] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759'. Checking compatability for channels 'SIP/zvonobot-00000029' and 'Recorder/ARI-00000036;2' [Aug 18 10:34:27] DEBUG[14394] bridge_native_rtp.c: Bridge 'fa4500bd-5aa0-4ef0-bd71-2993d47cd759' can not use native RTP bridge as channel 'SIP/zvonobot-00000029' has features which prevent it [Aug 18 10:34:27] DEBUG[14394] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[14394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14394] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14394] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[14394] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759 is already using the new technology. [Aug 18 10:34:27] DEBUG[14975] stasis.c: Topic 'bridge:3e9f0826-47ef-4258-a05a-53af5ce8577c': 0x7f0c100e1380 created [Aug 18 10:34:27] DEBUG[14999] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:27] DEBUG[14999] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213006&app=calls_0&format=slin16&external_host=127.0.0.1%3A50282 [Aug 18 10:34:27] DEBUG[14975] stasis.c: Creating topic. name: cache:636/bridge:3e9f0826-47ef-4258-a05a-53af5ce8577c, detail: [Aug 18 10:34:27] DEBUG[14975] stasis.c: Topic 'cache:636/bridge:3e9f0826-47ef-4258-a05a-53af5ce8577c': 0x7f0c100f6400 created [Aug 18 10:34:27] DEBUG[14975] bridge_native_rtp.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[14975] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[14975] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14975] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[14975] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[14975] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[14975] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[14998] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17184 RTP/AVP 0 8 101 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:27] DEBUG[14975] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:27] DEBUG[15000] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:27] DEBUG[15000] http.c: HTTP Request URI is /ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/addChannel?channel=213025 [Aug 18 10:34:27] DEBUG[15000] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 (Checking To) --From tag as54e004b1 --To-tag as58786f9e [Aug 18 10:34:27] DEBUG[15000] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15000] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/addChannel] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15000] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14975] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Finding handler for bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/addChannel [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Finding handler for 3e9f0826-47ef-4258-a05a-53af5ce8577c [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15000] res_ari.c: No explicit handler found for 3e9f0826-47ef-4258-a05a-53af5ce8577c. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Finding handler for addChannel [Aug 18 10:34:27] DEBUG[15000] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:27] DEBUG[14676] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[15000] stasis/control.c: 213025: Sending channel add_to_bridge command [Aug 18 10:34:27] DEBUG[14999] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14988] stasis/app.c: Channel '1629282863.496' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[14676] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:27] DEBUG[14998] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Stopping retransmission on '56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Got SDP version 1451248754 and unique parts [root 1451248754 IN IP4 178.62.121.41] [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1451248754 1451248754 IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c020490) ICE set role failed; no ice instance [Aug 18 10:34:27] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c020490) RTCP setting address on RTP instance [Aug 18 10:34:27] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c8c0246a0 -- Strict RTP learning after remote address set to: 178.62.121.41:17184 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:17184 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0043418) from 0x7f0c147e2330 to 0x7f0c8c020668 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb013da88) from 0x7f0c147e2330 to 0x7f0c8c020668 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb004ba88) from 0x7f0c147e2330 to 0x7f0c8c020668 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c020490) RTCP ignoring duplicate property [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002d setting read format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000002d setting write format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[14999] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c020490) DTLS - ast_rtp_activate rtp=0x7f0c8c0246a0 - setup and perform DTLS' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0246a0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c0246a0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:27] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Strict routing enforced for session 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117030@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f24ae17 Max-Forwards: 70 From: ;tag=as54e004b1 To: ;tag=as58786f9e Contact: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:27] DEBUG[14988] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 194 instead [Aug 18 10:34:27] DEBUG[14999] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[13415] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000003e [Aug 18 10:34:27] DEBUG[13415] stasis/control.c: 213025: Adding to bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c [Aug 18 10:34:27] DEBUG[13415] stasis/app.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[15001] bridge_channel.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: 0x7f0c84068b70(SIP/zvonobot-0000003e) is joining [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (1) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:27] DEBUG[14998] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session timer started: 20 - 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 1768000ms [Aug 18 10:34:27] VERBOSE[13192] dial.c: SIP/zvonobot-0000002d answered [Aug 18 10:34:27] VERBOSE[13192] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000002d [Aug 18 10:34:27] DEBUG[13192] stasis/app.c: Channel '213010' is 2 interested in calls_0 [Aug 18 10:34:27] VERBOSE[13192] res_rtp_asterisk.c: 0x7f0c8c0246a0 -- Strict RTP switching to RTP target address 178.62.121.41:17184 as source [Aug 18 10:34:27] DEBUG[13192] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:27] DEBUG[13192] channel.c: Channel SIP/zvonobot-0000002d setting read format path: ulaw -> alaw [Aug 18 10:34:27] DEBUG[13192] channel.c: Channel SIP/zvonobot-0000002d setting write format path: alaw -> ulaw [Aug 18 10:34:27] DEBUG[14672] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14999] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15001] bridge_channel.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: pushing 0x7f0c84068b70(SIP/zvonobot-0000003e) [Aug 18 10:34:27] VERBOSE[15001] bridge_channel.c: Channel SIP/zvonobot-0000003e joined 'simple_bridge' stasis-bridge <3e9f0826-47ef-4258-a05a-53af5ce8577c> [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:27] DEBUG[14672] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14998] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15001] bridge_native_rtp.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15001] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15001] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c is already using the new technology. [Aug 18 10:34:27] DEBUG[15002] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15001] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: 0x7f0c84068b70(SIP/zvonobot-0000003e) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[15002] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:27] DEBUG[15001] res_rtp_asterisk.c: (0x7f0c2c008d30) RTP changing ssrc from 1773489447 to 906331412 due to a source change [Aug 18 10:34:27] DEBUG[13415] stasis/app.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[15000] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] DEBUG[15000] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15002] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a;received=159.65.48.104 From: ;tag=as3ecc0b7c To: ;tag=as6b227143 Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66504757" Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[15002] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:27] DEBUG[15002] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15002] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:27] DEBUG[15003] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[14544] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14544] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK459afc0a;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3ecc0b7c [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6b227143 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14394] audiohook.c: Audiohook 0x7f0c8c14b960 has stale audio in its factories. Flushing them both [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15003] http.c: HTTP Request URI is /ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/record?name=213025_lNYdPoaZyNgkYDodNHVSYmzebqmBqduB&format=wav [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15002] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66504757" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 (Checking To) --From tag as3ecc0b7c --To-tag as6b227143 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 77ec81a43645f30730cc74c217742e98@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[14536] channel.c: Channel 0x7f0ca40f4750 'Snoop/212969-00000017' allocated [Aug 18 10:34:27] DEBUG[15003] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/record] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[14536] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[14295] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[14295] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[14295] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14295] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14295] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[14295] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d is already using the new technology. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2;received=159.65.48.104 From: ;tag=as1cc5f222 To: ;tag=as46a80351 Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259c04d7" Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1cc5f222 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as46a80351 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259c04d7" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:27] DEBUG[15002] stasis.c: Creating topic. name: bridge:0b66d66e-5f5c-4963-a022-79b61565f473, detail: [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 (Checking To) --From tag as1cc5f222 --To-tag as46a80351 [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Stopping retransmission on '517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116913@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2 Max-Forwards: 70 From: ;tag=as1cc5f222 To: Contact: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5 Max-Forwards: 70 From: ;tag=as6847ab41 To: ;tag=as0d63cc42 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="5cc7861d", response="2ddbce0d9077c405df851db6d16b3c6e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as6847ab41 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0d63cc42 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="5cc7861d", response="2ddbce0d9077c405df851db6d16b3c6e" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 (Checking From) --From tag as6847ab41 --To-tag as0d63cc42 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15003] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/record] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[14540] chan_sip.c: Hangup call SIP/zvonobot-000000a2, SIP callid 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14814] channel.c: Channel 0x7f0c08080020 'Recorder/ARI-0000004c;1' allocated [Aug 18 10:34:27] DEBUG[14814] stasis.c: Creating topic. name: channel:1629282867.557, detail: [Aug 18 10:34:27] DEBUG[14542] chan_sip.c: Hangup call SIP/zvonobot-000000a3, SIP callid 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14542] res_rtp_asterisk.c: (0x7f0ca40fbc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14542] res_rtp_asterisk.c: (0x7f0ca40fbc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14542] channel.c: Channel 0x7f0ca4122130 'SIP/zvonobot-000000a3' destroying [Aug 18 10:34:27] DEBUG[14534] chan_sip.c: Hangup call SIP/zvonobot-000000a0, SIP callid 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14534] res_rtp_asterisk.c: (0x7f0c8007aae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14534] res_rtp_asterisk.c: (0x7f0c8007aae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14534] channel.c: Channel 0x7f0c8009df10 'SIP/zvonobot-000000a0' destroying [Aug 18 10:34:27] DEBUG[14540] res_rtp_asterisk.c: (0x7f0c9c09bf60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[15003] http.c: match request [ari/bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/record] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14540] res_rtp_asterisk.c: (0x7f0c9c09bf60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14540] channel.c: Channel 0x7f0c9c0d6ff0 'SIP/zvonobot-000000a2' destroying [Aug 18 10:34:27] DEBUG[14535] chan_sip.c: Hangup call SIP/zvonobot-000000a1, SIP callid 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213129': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213129' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:413/channel:213129, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:413/channel:213129': 0x7f0ca4124930 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213129, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213129': 0x7f0ca4123eb0 destroyed [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213124': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213124' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213127': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213127' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:412/channel:213127, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:412/channel:213127': 0x7f0c9c025bd0 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213127, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213127': 0x7f0c9c0609e0 destroyed [Aug 18 10:34:27] DEBUG[14671] stasis.c: Creating topic. name: channel:1629282867.556, detail: [Aug 18 10:34:27] DEBUG[14671] stasis.c: Topic 'channel:1629282867.556': 0x7f0c940b15f0 created [Aug 18 10:34:27] DEBUG[14671] stasis.c: Creating topic. name: cache:637/channel:1629282867.556, detail: [Aug 18 10:34:27] DEBUG[14671] stasis.c: Topic 'cache:637/channel:1629282867.556': 0x7f0c940bb930 created [Aug 18 10:34:27] DEBUG[14709] channel.c: Channel 0x7f0c7c08a120 'Recorder/ARI-00000045;2' allocated [Aug 18 10:34:27] DEBUG[14709] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:27] DEBUG[14535] res_rtp_asterisk.c: (0x7f0ca0108a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14684] res_stasis_playback.c: 1629282853.352: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:27] DEBUG[15006] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15006] http.c: HTTP Request URI is /ari/channels/213129 [Aug 18 10:34:27] DEBUG[15006] http.c: match request [ari/channels/213129] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15006] http.c: match request [ari/channels/213129] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15006] http.c: match request [ari/channels/213129] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15006] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Finding handler for channels/213129 [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Finding handler for 213129 [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking channels create: Didn't match 213129 [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15006] res_ari.c: Checking channels externalMedia: Didn't match 213129 [Aug 18 10:34:27] DEBUG[15006] res_ari.c: No explicit handler found for 213129. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15008] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15008] http.c: HTTP Request URI is /ari/channels/213127 [Aug 18 10:34:27] DEBUG[15008] http.c: match request [ari/channels/213127] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15008] http.c: match request [ari/channels/213127] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15008] http.c: match request [ari/channels/213127] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15008] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Finding handler for channels/213127 [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Finding handler for 213127 [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking channels create: Didn't match 213127 [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15008] res_ari.c: Checking channels externalMedia: Didn't match 213127 [Aug 18 10:34:27] DEBUG[15008] res_ari.c: No explicit handler found for 213127. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[14684] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:27] DEBUG[14535] res_rtp_asterisk.c: (0x7f0ca0108a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14535] channel.c: Channel 0x7f0ca01129e0 'SIP/zvonobot-000000a1' destroying [Aug 18 10:34:27] DEBUG[15007] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[14684] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15002] stasis.c: Topic 'bridge:0b66d66e-5f5c-4963-a022-79b61565f473': 0x7f0c3401b3d0 created [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[15002] stasis.c: Creating topic. name: cache:638/bridge:0b66d66e-5f5c-4963-a022-79b61565f473, detail: [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:27] DEBUG[15007] http.c: HTTP Request URI is /ari/channels/213124 [Aug 18 10:34:27] DEBUG[15005] bridge_channel.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03: 0x7f0c7c04d110(Recorder/ARI-00000045;2) is joining [Aug 18 10:34:27] DEBUG[15005] bridge_channel.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03: pushing 0x7f0c7c04d110(Recorder/ARI-00000045;2) [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:27] DEBUG[15007] http.c: match request [ari/channels/213124] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.558, detail: [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 688, ms is 63 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213128': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213128' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[14998] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:27] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[14534] stasis.c: Destroying topic. name: cache:409/channel:213124, detail: [Aug 18 10:34:27] DEBUG[14534] stasis.c: Topic 'cache:409/channel:213124': 0x7f0c800a06e0 destroyed [Aug 18 10:34:27] DEBUG[14534] stasis.c: Destroying topic. name: channel:213124, detail: [Aug 18 10:34:27] DEBUG[14534] stasis.c: Topic 'channel:213124': 0x7f0c8009fc60 destroyed [Aug 18 10:34:27] DEBUG[15010] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15007] http.c: match request [ari/channels/213124] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15010] http.c: HTTP Request URI is /ari/channels/213128 [Aug 18 10:34:27] DEBUG[15002] stasis.c: Topic 'cache:638/bridge:0b66d66e-5f5c-4963-a022-79b61565f473': 0x7f0c34088a40 created [Aug 18 10:34:27] DEBUG[15002] bridge_native_rtp.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15002] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15002] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[15002] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15002] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15002] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK16d870c5;received=178.62.121.41 From: ;tag=as6847ab41 To: ;tag=as0d63cc42 Call-ID: 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475453 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790570 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (2) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3;received=159.65.48.104 From: ;tag=as7674a2b1 To: ;tag=as72f2af64 Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60d3994a" Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c6bccb3;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7674a2b1 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as72f2af64 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60d3994a" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 (Checking To) --From tag as7674a2b1 --To-tag as72f2af64 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4210fe0406a05abe76ae857d60be22d3@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca40fbc00) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca40fbc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca40fbc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca40fbc00) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca40fbc00' [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2749fa7d41ec862f1556002a63546011@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8007aae0) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8007aae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8007aae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8007aae0) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8007aae0' [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c09bf60) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c09bf60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c09bf60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c09bf60) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9c09bf60' [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '79bc8168684fc820560b1e9c5fec00f7@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca0108a20) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca0108a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca0108a20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca0108a20) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca0108a20' [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424 Max-Forwards: 70 From: ;tag=as0d3ccf68 To: ;tag=as4d3d785f Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="29c76a44", response="e2ffe165291edb147174178f90098442" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as0d3ccf68 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as4d3d785f [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="29c76a44", response="e2ffe165291edb147174178f90098442" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 (Checking From) --From tag as0d3ccf68 --To-tag as4d3d785f [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[14999] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:27] DEBUG[14999] netsock2.c: Splitting '127.0.0.1:50282' into... [Aug 18 10:34:27] DEBUG[14999] netsock2.c: ...host '127.0.0.1' and port '50282'. [Aug 18 10:34:27] DEBUG[14999] netsock2.c: Splitting '127.0.0.1:50282' into... [Aug 18 10:34:27] DEBUG[14999] netsock2.c: ...host '127.0.0.1' and port '50282'. [Aug 18 10:34:27] DEBUG[14999] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:27] DEBUG[14999] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c200c10c0' [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) RTP allocated port 11156 [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) ICE creating session 127.0.0.1:11156 (11156) [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) ICE create [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) ICE add system candidates [Aug 18 10:34:27] DEBUG[14999] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:27] DEBUG[14999] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) ICE add candidate: 159.65.48.104:11156, 2130706431 [Aug 18 10:34:27] DEBUG[14999] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:27] DEBUG[14999] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:27] DEBUG[14999] res_rtp_asterisk.c: (0x7f0c200c10c0) ICE add candidate: 10.131.0.10:11156, 2130706431 [Aug 18 10:34:27] DEBUG[14999] rtp_engine.c: RTP instance '0x7f0c200c10c0' is setup and ready to go [Aug 18 10:34:27] DEBUG[14999] stasis.c: Creating topic. name: channel:robot_213006, detail: [Aug 18 10:34:27] DEBUG[14999] stasis.c: Topic 'channel:robot_213006': 0x7f0c20011f50 created [Aug 18 10:34:27] DEBUG[14999] stasis.c: Creating topic. name: cache:639/channel:robot_213006, detail: [Aug 18 10:34:27] DEBUG[14999] stasis.c: Topic 'cache:639/channel:robot_213006': 0x7f0c20012940 created [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15007] http.c: match request [ari/channels/213124] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14814] stasis.c: Topic 'channel:1629282867.557': 0x7f0c08074300 created [Aug 18 10:34:27] DEBUG[14814] stasis.c: Creating topic. name: cache:640/channel:1629282867.557, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:411/channel:213128, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:411/channel:213128': 0x7f0ca00fd6c0 destroyed [Aug 18 10:34:27] DEBUG[14998] netsock2.c: Splitting '127.0.0.1:50312' into... [Aug 18 10:34:27] DEBUG[15002] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15002] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15007] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.558': 0x7f0c30113890 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:641/channel:1629282867.558, detail: [Aug 18 10:34:27] DEBUG[15003] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14998] netsock2.c: ...host '127.0.0.1' and port '50312'. [Aug 18 10:34:27] DEBUG[14998] netsock2.c: Splitting '127.0.0.1:50312' into... [Aug 18 10:34:27] DEBUG[14998] netsock2.c: ...host '127.0.0.1' and port '50312'. [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213128, detail: [Aug 18 10:34:27] DEBUG[14998] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:27] DEBUG[14998] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c240eada0' [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) RTP allocated port 19878 [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) ICE creating session 127.0.0.1:19878 (19878) [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) ICE create [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) ICE add system candidates [Aug 18 10:34:27] DEBUG[14998] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:27] DEBUG[14998] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) ICE add candidate: 159.65.48.104:19878, 2130706431 [Aug 18 10:34:27] DEBUG[14998] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:27] DEBUG[14998] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:27] DEBUG[14998] res_rtp_asterisk.c: (0x7f0c240eada0) ICE add candidate: 10.131.0.10:19878, 2130706431 [Aug 18 10:34:27] DEBUG[14998] rtp_engine.c: RTP instance '0x7f0c240eada0' is setup and ready to go [Aug 18 10:34:27] DEBUG[14998] stasis.c: Creating topic. name: channel:robot_212996, detail: [Aug 18 10:34:27] DEBUG[14814] stasis.c: Topic 'cache:640/channel:1629282867.557': 0x7f0c0802cbd0 created [Aug 18 10:34:27] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:641/channel:1629282867.558': 0x7f0c3007f1a0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:641/channel:1629282867.558, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:641/channel:1629282867.558': 0x7f0c3007f1a0 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213128': 0x7f0ca00767f0 destroyed [Aug 18 10:34:27] DEBUG[15004] stasis/app.c: Channel '1629282863.497' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[15004] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 199 instead [Aug 18 10:34:27] DEBUG[14657] channel.c: Channel 0x7f0c7804c540 'Recorder/ARI-00000034;1' destroying [Aug 18 10:34:27] DEBUG[15011] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[14998] stasis.c: Topic 'channel:robot_212996': 0x7f0c241253e0 created [Aug 18 10:34:27] DEBUG[14998] stasis.c: Creating topic. name: cache:642/channel:robot_212996, detail: [Aug 18 10:34:27] DEBUG[14998] stasis.c: Topic 'cache:642/channel:robot_212996': 0x7f0c24017710 created [Aug 18 10:34:27] DEBUG[15009] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[13221] res_rtp_asterisk.c: (0x7f0c28011240) DTLS stop [Aug 18 10:34:27] DEBUG[13221] res_rtp_asterisk.c: (0x7f0c28011240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[13221] res_rtp_asterisk.c: (0x7f0c28011240) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[13221] res_rtp_asterisk.c: (0x7f0c28011240) ICE stopped [Aug 18 10:34:27] DEBUG[13221] rtp_engine.c: Destroyed RTP instance '0x7f0c28011240' [Aug 18 10:34:27] DEBUG[13221] channel.c: Channel 0x7f0c2807fb90 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240' destroying [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Finding handler for channels/213124 [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Finding handler for 213124 [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking channels create: Didn't match 213124 [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15010] http.c: match request [ari/channels/213128] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15005] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:27] VERBOSE[15005] bridge_channel.c: Channel Recorder/ARI-00000045;2 joined 'simple_bridge' stasis-bridge <804476f3-9df1-4495-8c76-b406f7162d03> [Aug 18 10:34:27] DEBUG[15005] bridge_native_rtp.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03'. Checking compatability for channels 'SIP/zvonobot-0000004c' and 'Recorder/ARI-00000045;2' [Aug 18 10:34:27] DEBUG[15005] bridge_native_rtp.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03' can not use native RTP bridge as could not get details [Aug 18 10:34:27] DEBUG[15005] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15005] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15005] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15005] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15005] bridge.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03 is already using the new technology. [Aug 18 10:34:27] DEBUG[15005] bridge.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03: 0x7f0c7c04d110(Recorder/ARI-00000045;2) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel Recorder/ARI-00000045;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel SIP/zvonobot-0000004c setting write format path: slin -> ulaw [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel SIP/zvonobot-0000004c setting read format path: ulaw -> slin [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel Recorder/ARI-00000045;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[14813] stasis.c: Creating topic. name: channel:1629282867.561, detail: [Aug 18 10:34:27] DEBUG[14713] channel.c: Channel 0x7f0c8c03b720 'Recorder/ARI-00000046;2' allocated [Aug 18 10:34:27] DEBUG[14713] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:27] DEBUG[14813] stasis.c: Topic 'channel:1629282867.561': 0x7f0c10103cf0 created [Aug 18 10:34:27] DEBUG[14813] stasis.c: Creating topic. name: cache:643/channel:1629282867.561, detail: [Aug 18 10:34:27] DEBUG[14813] stasis.c: Topic 'cache:643/channel:1629282867.561': 0x7f0c10003e70 created [Aug 18 10:34:27] DEBUG[14654] bridge_channel.c: Setting 0x7f0c780664b0(Recorder/ARI-00000034;2) state from:0 to:1 [Aug 18 10:34:27] DEBUG[14654] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: pulling 0x7f0c780664b0(Recorder/ARI-00000034;2) [Aug 18 10:34:27] VERBOSE[14654] bridge_channel.c: Channel Recorder/ARI-00000034;2 left 'simple_bridge' stasis-bridge <94ef4fc9-246b-4999-9567-b41f8ba44681> [Aug 18 10:34:27] DEBUG[14654] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c780664b0(Recorder/ARI-00000034;2) is leaving simple_bridge technology [Aug 18 10:34:27] DEBUG[14654] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[14654] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[14654] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14654] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[14654] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[14654] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 is already using the new technology. [Aug 18 10:34:27] DEBUG[15011] http.c: HTTP Request URI is /ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/addChannel?channel=213010 [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] DEBUG[14890] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:27] DEBUG[15009] http.c: HTTP Request URI is /ari/playbacks/3e5b571e-c10e-4264-a2cf-00bb76979c7b [Aug 18 10:34:27] DEBUG[15007] res_ari.c: Checking channels externalMedia: Didn't match 213124 [Aug 18 10:34:27] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15010] http.c: match request [ari/channels/213128] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14657] stasis.c: Destroying topic. name: cache:366/channel:1629282851.317, detail: [Aug 18 10:34:27] DEBUG[14657] stasis.c: Topic 'cache:366/channel:1629282851.317': 0x7f0c78053060 destroyed [Aug 18 10:34:27] DEBUG[14657] stasis.c: Destroying topic. name: channel:1629282851.317, detail: [Aug 18 10:34:27] DEBUG[14657] stasis.c: Topic 'channel:1629282851.317': 0x7f0c7801a240 destroyed [Aug 18 10:34:27] DEBUG[14654] channel.c: Channel 0x7f0c78069a80 'Recorder/ARI-00000034;2' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[15007] res_ari.c: No explicit handler found for 213124. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[15010] http.c: match request [ari/channels/213128] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel 'robot_212983': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel 'robot_212983' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:75/channel:robot_212983, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:75/channel:robot_212983': 0x7f0c2807d180 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212983, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:robot_212983': 0x7f0c2800a760 destroyed [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15012] bridge_channel.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) is joining [Aug 18 10:34:27] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Finding handler for bridges/3e9f0826-47ef-4258-a05a-53af5ce8577c/record [Aug 18 10:34:27] DEBUG[13351] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.558, detail: [Aug 18 10:34:27] DEBUG[15009] http.c: match request [ari/playbacks/3e5b571e-c10e-4264-a2cf-00bb76979c7b] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15010] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15011] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[13466] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15009] http.c: match request [ari/playbacks/3e5b571e-c10e-4264-a2cf-00bb76979c7b] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.558': 0x7f0c30113890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a3', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213129', '')] [Aug 18 10:34:27] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15009] http.c: match request [ari/playbacks/3e5b571e-c10e-4264-a2cf-00bb76979c7b] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[15009] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Finding handler for playbacks/3e5b571e-c10e-4264-a2cf-00bb76979c7b [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Finding handler for playbacks [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:27] DEBUG[13327] res_rtp_asterisk.c: (0x7f0cb0015bc0) RTP no remote address on instance, so dropping frame [Aug 18 10:34:27] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Finding handler for channels/213128 [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Finding handler for 213128 [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking channels create: Didn't match 213128 [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15010] res_ari.c: Checking channels externalMedia: Didn't match 213128 [Aug 18 10:34:27] DEBUG[15010] res_ari.c: No explicit handler found for 213128. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15012] bridge_channel.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: pushing 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) [Aug 18 10:34:27] DEBUG[15011] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[14709] res_stasis_recording.c: 1629282860.447: Sending record(213041_SDPjzwBTGJBwjIEyyHVbOCOACYofXdxs.wav) command [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424;received=178.62.121.41 From: ;tag=as0d3ccf68 To: ;tag=as4d3d785f Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612348 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (2) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116881@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1e3fcdcc Max-Forwards: 70 From: ;tag=as7eb8ca07 To: Contact: Call-ID: 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2026655962 2026655962 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11948 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session timer stopped: 5 - 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c14d3a0;received=159.65.48.104 From: ;tag=as22df0306 To: ;tag=as67bc1c22 Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 861929435 861929435 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10478 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6c14d3a0;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22df0306 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as67bc1c22 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.562, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.562': 0x7f0c30162ad0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:644/channel:1629282867.562, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:644/channel:1629282867.562': 0x7f0c300ff110 created [Aug 18 10:34:27] DEBUG[14709] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Finding handler for 3e9f0826-47ef-4258-a05a-53af5ce8577c [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15003] res_ari.c: No explicit handler found for 3e9f0826-47ef-4258-a05a-53af5ce8577c. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Finding handler for record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:27] DEBUG[15003] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:27] DEBUG[15003] stasis.c: Creating topic. name: channel:1629282867.563, detail: [Aug 18 10:34:27] DEBUG[14709] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15003] stasis.c: Topic 'channel:1629282867.563': 0x7f0c300b1880 created [Aug 18 10:34:27] DEBUG[15003] stasis.c: Creating topic. name: cache:645/channel:1629282867.563, detail: [Aug 18 10:34:27] DEBUG[15003] stasis.c: Topic 'cache:645/channel:1629282867.563': 0x7f0c300d6230 created [Aug 18 10:34:27] DEBUG[15014] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15014] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:27] DEBUG[15014] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15014] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15014] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15014] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15011] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/addChannel] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:27] DEBUG[15011] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:644/channel:1629282867.562, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:644/channel:1629282867.562': 0x7f0c300ff110 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.562, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.562': 0x7f0c30162ad0 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:16', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a0', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213124', '')] [Aug 18 10:34:27] VERBOSE[14799] res_rtp_asterisk.c: 0x7f0ca000e480 -- Strict RTP learning complete - Locking on source address 178.62.121.41:15546 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 861929435 861929435 IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Finding handler for bridges/0b66d66e-5f5c-4963-a022-79b61565f473/addChannel [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15013] app.c: play_and_record: , /var/spool/asterisk/recording/213041_SDPjzwBTGJBwjIEyyHVbOCOACYofXdxs, 'wav' [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Setting 0x7f0cac01e580(SIP/zvonobot-00000010) state from:0 to:1 [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Finding handler for 0b66d66e-5f5c-4963-a022-79b61565f473 [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15011] res_ari.c: No explicit handler found for 0b66d66e-5f5c-4963-a022-79b61565f473. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Finding handler for addChannel [Aug 18 10:34:27] DEBUG[15011] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:27] DEBUG[15011] stasis/control.c: 213010: Sending channel add_to_bridge command [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10478 RTP/AVP 0 8 101 [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel Recorder/ARI-00000045;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:27] DEBUG[13192] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000002d [Aug 18 10:34:27] DEBUG[13192] stasis/control.c: 213010: Adding to bridge 0b66d66e-5f5c-4963-a022-79b61565f473 [Aug 18 10:34:27] DEBUG[13192] stasis/app.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:27] DEBUG[15013] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:27] DEBUG[15012] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:27] VERBOSE[15012] bridge_channel.c: Channel Recorder/ARI-00000046;2 joined 'simple_bridge' stasis-bridge <6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb> [Aug 18 10:34:27] DEBUG[14719] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14899] app.c: One waitfor failed, trying another [Aug 18 10:34:27] DEBUG[15005] channel.c: Channel Recorder/ARI-00000045;2 setting write format path: alaw -> slin [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 (Checking To) --From tag as22df0306 --To-tag as67bc1c22 [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Finding handler for 3e5b571e-c10e-4264-a2cf-00bb76979c7b [Aug 18 10:34:27] DEBUG[15015] bridge_channel.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: 0x7f0c8808d730(SIP/zvonobot-0000002d) is joining [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Stopping retransmission on '72b609e44febd02a23174a4d311879ff@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Got SDP version 861929435 and unique parts [root 861929435 IN IP4 178.62.121.41] [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 861929435 861929435 IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) ICE set role failed; no ice instance [Aug 18 10:34:27] DEBUG[13342] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[14719] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) RTCP setting address on RTP instance [Aug 18 10:34:27] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c3c04ba20 -- Strict RTP learning after remote address set to: 178.62.121.41:10478 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:10478 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb003fa78) from 0x7f0c147e2330 to 0x7f0c3c032008 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0024cb8) from 0x7f0c147e2330 to 0x7f0c3c032008 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00973f8) from 0x7f0c147e2330 to 0x7f0c3c032008 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) RTCP ignoring duplicate property [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000044 setting read format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000044 setting write format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) DTLS - ast_rtp_activate rtp=0x7f0c3c04ba20 - setup and perform DTLS' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c04ba20) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c04ba20) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:27] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Strict routing enforced for session 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117012@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK70d7785a Max-Forwards: 70 From: ;tag=as22df0306 To: ;tag=as67bc1c22 Contact: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[15015] bridge_channel.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: pushing 0x7f0c8808d730(SIP/zvonobot-0000002d) [Aug 18 10:34:27] VERBOSE[15015] bridge_channel.c: Channel SIP/zvonobot-0000002d joined 'simple_bridge' stasis-bridge <0b66d66e-5f5c-4963-a022-79b61565f473> [Aug 18 10:34:27] DEBUG[15015] bridge_native_rtp.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15015] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15015] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15015] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15015] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15015] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473 is already using the new technology. [Aug 18 10:34:27] DEBUG[15015] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: 0x7f0c8808d730(SIP/zvonobot-0000002d) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[15015] res_rtp_asterisk.c: (0x7f0c8c020490) RTP changing ssrc from 223112201 to 248771747 due to a source change [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[14799] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTCP got report of 76 bytes from 178.62.121.41:15547 [Aug 18 10:34:27] VERBOSE[13426] dial.c: SIP/zvonobot-00000044 answered [Aug 18 10:34:27] VERBOSE[13426] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000044 [Aug 18 10:34:27] DEBUG[13426] stasis/app.c: Channel '213028' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[14718] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14718] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15012] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb'. Checking compatability for channels 'SIP/zvonobot-0000003c' and 'Recorder/ARI-00000046;2' [Aug 18 10:34:27] DEBUG[15012] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb' can not use native RTP bridge as could not get details [Aug 18 10:34:27] DEBUG[15012] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15012] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15012] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15012] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15012] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb is already using the new technology. [Aug 18 10:34:27] DEBUG[15012] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[15012] channel.c: Channel Recorder/ARI-00000046;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[15012] channel.c: Channel SIP/zvonobot-0000003c setting write format path: slin -> alaw [Aug 18 10:34:27] DEBUG[15012] channel.c: Channel SIP/zvonobot-0000003c setting read format path: alaw -> slin [Aug 18 10:34:27] DEBUG[15012] channel.c: Channel Recorder/ARI-00000046;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[15009] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:27] DEBUG[14713] res_stasis_recording.c: 1629282860.450: Sending record(213027_LzhbtVqcAVRxNKmskAtSCBZXSwnXqDKu.wav) command [Aug 18 10:34:27] DEBUG[15011] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[14713] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:27] DEBUG[14713] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15009] res_ari.c: No explicit handler found for 3e5b571e-c10e-4264-a2cf-00bb76979c7b. Using wildcard playbackId. [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.564, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.564': 0x7f0c3013cd20 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:646/channel:1629282867.564, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:646/channel:1629282867.564': 0x7f0c3013cdd0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:646/channel:1629282867.564, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:646/channel:1629282867.564': 0x7f0c3013cdd0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.564, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.564': 0x7f0c3013cd20 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a2', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213127', '')] [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15011] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[13192] stasis/app.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[15009] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] DEBUG[15009] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session timer started: 4 - 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 1768000ms [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.565, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.565': 0x7f0c301139c0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:647/channel:1629282867.565, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:647/channel:1629282867.565': 0x7f0c301645d0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:647/channel:1629282867.565, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:647/channel:1629282867.565': 0x7f0c301645d0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.565, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.565': 0x7f0c301139c0 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a1', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213128', '')] [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pulling 0x7f0cac01e580(SIP/zvonobot-00000010) [Aug 18 10:34:27] VERBOSE[13327] bridge_channel.c: Channel SIP/zvonobot-00000010 left 'softmix' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:34:27] DEBUG[15018] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0cac01e580(SIP/zvonobot-00000010) is leaving softmix technology [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Setting 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) state from:0 to:2 [Aug 18 10:34:27] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[15019] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15018] http.c: HTTP Request URI is /ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/record?name=213010_RLEHAKgKjztapoIEIJWLtEnzYYryiZuH&format=wav [Aug 18 10:34:27] DEBUG[15018] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/record] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15018] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/record] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15018] http.c: match request [ari/bridges/0b66d66e-5f5c-4963-a022-79b61565f473/record] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15018] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Finding handler for bridges/0b66d66e-5f5c-4963-a022-79b61565f473/record [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15016] app.c: play_and_record: , /var/spool/asterisk/recording/213027_LzhbtVqcAVRxNKmskAtSCBZXSwnXqDKu, 'wav' [Aug 18 10:34:27] DEBUG[15019] http.c: HTTP Request URI is /ari/channels/robot_213009 [Aug 18 10:34:27] VERBOSE[15013] app.c: x=0, open writing: /var/spool/asterisk/recording/213041_SDPjzwBTGJBwjIEyyHVbOCOACYofXdxs format: wav, 0x7f0c8c01cb70 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc;received=159.65.48.104 From: ;tag=as2a62315e To: ;tag=as3efd5c26 Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42031c6d" Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5cee45bc;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2a62315e [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3efd5c26 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42031c6d" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 (Checking To) --From tag as2a62315e --To-tag as3efd5c26 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7f1b1c696a88fe9a2e47206a14e70f07@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6255ms with no response [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Hanging up call 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15019] http.c: match request [ari/channels/robot_213009] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15016] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.566, detail: [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15017] http.c: HTTP opening session. Top level [Aug 18 10:34:27] VERBOSE[15016] app.c: x=0, open writing: /var/spool/asterisk/recording/213027_LzhbtVqcAVRxNKmskAtSCBZXSwnXqDKu format: wav, 0x7f0c9007f790 [Aug 18 10:34:27] DEBUG[13627] audiohook.c: Audiohook 0x7f0c180dbfa0 has stale audio in its factories. Flushing them both [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.566': 0x7f0c30070890 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:648/channel:1629282867.566, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:648/channel:1629282867.566': 0x7f0c300d6d00 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:648/channel:1629282867.566, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:648/channel:1629282867.566': 0x7f0c300d6d00 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.566, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.566': 0x7f0c30070890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50397-0x7f0c28011240', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212983', '')] [Aug 18 10:34:27] DEBUG[15019] http.c: match request [ari/channels/robot_213009] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15014] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[14721] channel.c: Soft-Hanging (0x20) up channel 'SIP/zvonobot-0000003b' [Aug 18 10:34:27] DEBUG[15017] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Finding handler for 0b66d66e-5f5c-4963-a022-79b61565f473 [Aug 18 10:34:27] DEBUG[14736] channel.c: Channel 0x7f0c1c09dfb0 'SIP/zvonobot-000000b4' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[15019] http.c: match request [ari/channels/robot_213009] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[14721] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] DEBUG[14721] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:27] DEBUG[13346] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15018] res_ari.c: No explicit handler found for 0b66d66e-5f5c-4963-a022-79b61565f473. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15017] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15014] stasis.c: Creating topic. name: bridge:c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4, detail: [Aug 18 10:34:27] DEBUG[15014] stasis.c: Topic 'bridge:c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4': 0x7f0c88011260 created [Aug 18 10:34:27] DEBUG[15014] stasis.c: Creating topic. name: cache:649/bridge:c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4, detail: [Aug 18 10:34:27] DEBUG[15014] stasis.c: Topic 'cache:649/bridge:c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4': 0x7f0c88051a00 created [Aug 18 10:34:27] DEBUG[15014] bridge_native_rtp.c: Bridge 'c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15014] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15014] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15014] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[15014] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15014] bridge.c: Bridge c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15014] bridge.c: Bridge c4d7ec35-5fff-46e7-8b4e-8d7bfe7534e4: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[15021] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15021] http.c: HTTP Request URI is /ari/channels/213041/snoop?app=calls_0&spy=in [Aug 18 10:34:27] DEBUG[15021] http.c: match request [ari/channels/213041/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15021] http.c: match request [ari/channels/213041/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15021] http.c: match request [ari/channels/213041/snoop] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15021] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Finding handler for channels/213041/snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[13327] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c'. Checking compatability for channels 'Announcer/ARI-0000000b;2' and 'Recorder/ARI-0000000a;2' [Aug 18 10:34:27] DEBUG[13327] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as could not get details [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13327] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] VERBOSE[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: switching from softmix technology to simple_bridge [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: moving 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) to dummy bridge temporarily [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: moving 0x7f0c88048a20(Recorder/ARI-0000000a;2) to dummy bridge temporarily [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) is leaving softmix technology (dummy) [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is leaving softmix technology (dummy) [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling softmix technology stop [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Recorder/ARI-0000000a;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Announcer/ARI-0000000b;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Announcer/ARI-0000000b;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Recorder/ARI-0000000a;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Recorder/ARI-0000000a;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Announcer/ARI-0000000b;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Announcer/ARI-0000000b;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Finding handler for record [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[15019] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Setting 0x7f0ca803dbf0(SIP/zvonobot-0000003b) state from:0 to:1 [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel Recorder/ARI-0000000a;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:27] DEBUG[15022] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15020] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15022] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Finding handler for channels/robot_213009 [Aug 18 10:34:27] DEBUG[15020] http.c: HTTP Request URI is /ari/channels/1629282840.199 [Aug 18 10:34:27] DEBUG[15022] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15020] http.c: match request [ari/channels/1629282840.199] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15022] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15020] http.c: match request [ari/channels/1629282840.199] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15022] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15020] http.c: match request [ari/channels/1629282840.199] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15014] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15022] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:27] DEBUG[15020] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15017] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Finding handler for robot_213009 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking channels create: Didn't match robot_213009 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15019] res_ari.c: Checking channels externalMedia: Didn't match robot_213009 [Aug 18 10:34:27] DEBUG[15019] res_ari.c: No explicit handler found for robot_213009. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Finding handler for channels/1629282840.199 [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Finding handler for 1629282840.199 [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking channels create: Didn't match 1629282840.199 [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15020] res_ari.c: Checking channels externalMedia: Didn't match 1629282840.199 [Aug 18 10:34:27] DEBUG[15020] res_ari.c: No explicit handler found for 1629282840.199. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:27] DEBUG[15018] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:27] DEBUG[15017] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15017] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15018] stasis.c: Creating topic. name: channel:1629282867.567, detail: [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350 Max-Forwards: 70 From: ;tag=as1f220605 To: ;tag=as0b424b33 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1ea6bc94", response="1600c32e40071d00c5ebdbecee32a250" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350 [Aug 18 10:34:27] DEBUG[15014] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: deferring softmix technology destructor [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as1f220605 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as0b424b33 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1ea6bc94", response="1600c32e40071d00c5ebdbecee32a250" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 (Checking From) --From tag as1f220605 --To-tag as0b424b33 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[13327] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: queueing action type:13 sub:1000 [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:27] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15018] stasis.c: Topic 'channel:1629282867.567': 0x7f0c9c0d2f50 created [Aug 18 10:34:27] DEBUG[15018] stasis.c: Creating topic. name: cache:650/channel:1629282867.567, detail: [Aug 18 10:34:27] DEBUG[15018] stasis.c: Topic 'cache:650/channel:1629282867.567': 0x7f0c9c03a160 created [Aug 18 10:34:27] DEBUG[13695] audiohook.c: Audiohook 0x7f0cb011ab60 has stale audio in its factories. Flushing them both [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Finding handler for 213041 [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channels create: Didn't match 213041 [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b4 - start 1629282860.964342 answer 0.000000 end 1629282867.448188 dur 6.483 bill 1629282867.448 dispo NO ANSWER [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000010 - start 1629282824.163123 answer 1629282833.179731 end 1629282867.466246 dur 43.303 bill 34.286 dispo ANSWERED [Aug 18 10:34:27] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channels externalMedia: Didn't match 213041 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1780e350;received=178.62.121.41 From: ;tag=as1f220605 To: ;tag=as0b424b33 Call-ID: 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6235ms with no response [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Hanging up call 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15021] res_ari.c: No explicit handler found for 213041. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[14739] channel.c: Channel 0x7f0cb4024d90 'SIP/zvonobot-000000b5' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pulling 0x7f0ca803dbf0(SIP/zvonobot-0000003b) [Aug 18 10:34:27] VERBOSE[13627] bridge_channel.c: Channel SIP/zvonobot-0000003b left 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Finding handler for snoop [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0ca803dbf0(SIP/zvonobot-0000003b) is leaving simple_bridge technology [Aug 18 10:34:27] DEBUG[13632] app.c: One waitfor failed, trying another [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Setting 0x7f0c94055bb0(Recorder/ARI-0000001e;2) state from:0 to:2 [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:27] DEBUG[15021] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:27] DEBUG[15022] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[13342] bridge_softmix.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: stopping mixing thread [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Bridge is returning 0x7f0cac01e580(SIP/zvonobot-00000010) to read format alaw [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel SIP/zvonobot-00000010 setting read format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[13327] bridge_channel.c: Bridge is returning 0x7f0cac01e580(SIP/zvonobot-00000010) to write format alaw [Aug 18 10:34:27] DEBUG[13327] channel.c: Channel SIP/zvonobot-00000010 setting write format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[13327] stasis/control.c: 212981, 0ec77f0c-7a86-4072-a1c4-e42f5256208c: Channel was departed from bridge [Aug 18 10:34:27] DEBUG[13327] stasis/app.c: bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c': is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[12962] stasis/control.c: 212981: Channel departing bridge [Aug 18 10:34:27] DEBUG[12962] bridge.c: Waiting for 0x7f0cac01e580(SIP/zvonobot-00000010) bridge thread to die. [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b5 - start 1629282861.128119 answer 0.000000 end 1629282867.550057 dur 6.421 bill 1629282867.550 dispo NO ANSWER [Aug 18 10:34:27] DEBUG[13340] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pulling 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) [Aug 18 10:34:27] DEBUG[13327] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] VERBOSE[13340] bridge_channel.c: Channel Announcer/ARI-0000000b;2 left 'simple_bridge' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:34:27] DEBUG[12962] stasis/app.c: channel '212981': is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[12962] channel.c: Channel 0x7f0cb003b730 'SIP/zvonobot-00000010' hanging up. Refs: 3 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:27] DEBUG[15022] stasis.c: Creating topic. name: bridge:413b28bc-b121-462c-8ad3-b989ef736d5a, detail: [Aug 18 10:34:27] DEBUG[15022] stasis.c: Topic 'bridge:413b28bc-b121-462c-8ad3-b989ef736d5a': 0x7f0cb00524d0 created [Aug 18 10:34:27] DEBUG[15022] stasis.c: Creating topic. name: cache:651/bridge:413b28bc-b121-462c-8ad3-b989ef736d5a, detail: [Aug 18 10:34:27] DEBUG[15022] stasis.c: Topic 'cache:651/bridge:413b28bc-b121-462c-8ad3-b989ef736d5a': 0x7f0cb0097550 created [Aug 18 10:34:27] DEBUG[15022] bridge_native_rtp.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15022] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15022] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15022] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[15022] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15022] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15022] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15017] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15017] stasis.c: Creating topic. name: bridge:32884ae9-cc8b-4356-9f35-b03521fb3a1e, detail: [Aug 18 10:34:27] DEBUG[15017] stasis.c: Topic 'bridge:32884ae9-cc8b-4356-9f35-b03521fb3a1e': 0x7f0ca8056f40 created [Aug 18 10:34:27] DEBUG[15017] stasis.c: Creating topic. name: cache:652/bridge:32884ae9-cc8b-4356-9f35-b03521fb3a1e, detail: [Aug 18 10:34:27] DEBUG[15017] stasis.c: Topic 'cache:652/bridge:32884ae9-cc8b-4356-9f35-b03521fb3a1e': 0x7f0ca80e86c0 created [Aug 18 10:34:27] DEBUG[13340] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0ca004e1f0(Announcer/ARI-0000000b;2) is leaving simple_bridge technology [Aug 18 10:34:27] DEBUG[13340] bridge_channel.c: Setting 0x7f0c88048a20(Recorder/ARI-0000000a;2) state from:0 to:2 [Aug 18 10:34:27] DEBUG[14838] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14555] chan_sip.c: Hangup call SIP/zvonobot-000000a8, SIP callid 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14555] res_rtp_asterisk.c: (0x2c86070) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14555] res_rtp_asterisk.c: (0x2c86070) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14555] channel.c: Channel 0x2c522e0 'SIP/zvonobot-000000a8' destroying [Aug 18 10:34:27] DEBUG[14554] chan_sip.c: Hangup call SIP/zvonobot-000000a7, SIP callid 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14554] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:27] DEBUG[14554] res_rtp_asterisk.c: (0x7f0c88038c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14554] res_rtp_asterisk.c: (0x7f0c88038c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14554] channel.c: Channel 0x7f0c88058300 'SIP/zvonobot-000000a7' destroying [Aug 18 10:34:27] DEBUG[14838] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14549] chan_sip.c: Hangup call SIP/zvonobot-000000a5, SIP callid 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14549] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:27] DEBUG[14549] res_rtp_asterisk.c: (0x7f0ca80ebb80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[13754] channel.c: Channel 0x7f0c7806bc10 'Recorder/ARI-0000001c;1' destroying [Aug 18 10:34:27] DEBUG[14549] res_rtp_asterisk.c: (0x7f0ca80ebb80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14549] channel.c: Channel 0x7f0ca8109f70 'SIP/zvonobot-000000a5' destroying [Aug 18 10:34:27] DEBUG[13340] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[13340] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[13340] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13340] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13340] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[13340] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c is already using the new technology. [Aug 18 10:34:27] DEBUG[20534] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:27] DEBUG[14988] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[14853] channel.c: Channel 0x7f0c9800e3c0 'Recorder/ARI-0000004d;1' allocated [Aug 18 10:34:27] DEBUG[20534] bridge_softmix.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: Waiting for mixing thread to die. [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[14853] stasis.c: Creating topic. name: channel:1629282867.568, detail: [Aug 18 10:34:27] DEBUG[13329] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: pulling 0x7f0c88048a20(Recorder/ARI-0000000a;2) [Aug 18 10:34:27] VERBOSE[13329] bridge_channel.c: Channel Recorder/ARI-0000000a;2 left 'simple_bridge' stasis-bridge <0ec77f0c-7a86-4072-a1c4-e42f5256208c> [Aug 18 10:34:27] DEBUG[13329] bridge_channel.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c: 0x7f0c88048a20(Recorder/ARI-0000000a;2) is leaving simple_bridge technology [Aug 18 10:34:27] DEBUG[13329] bridge_native_rtp.c: Bridge '0ec77f0c-7a86-4072-a1c4-e42f5256208c' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[13329] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[13329] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13329] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13329] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[13329] bridge.c: Bridge 0ec77f0c-7a86-4072-a1c4-e42f5256208c is already using the new technology. [Aug 18 10:34:27] DEBUG[14853] stasis.c: Topic 'channel:1629282867.568': 0x7f0c9809d960 created [Aug 18 10:34:27] DEBUG[14853] stasis.c: Creating topic. name: cache:653/channel:1629282867.568, detail: [Aug 18 10:34:27] DEBUG[14853] stasis.c: Topic 'cache:653/channel:1629282867.568': 0x7f0c980ad4a0 created [Aug 18 10:34:27] DEBUG[14734] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[13678] bridge_channel.c: Setting 0x7f0c78074930(Recorder/ARI-0000001c;2) state from:0 to:1 [Aug 18 10:34:27] DEBUG[13678] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pulling 0x7f0c78074930(Recorder/ARI-0000001c;2) [Aug 18 10:34:27] VERBOSE[13678] bridge_channel.c: Channel Recorder/ARI-0000001c;2 left 'softmix' stasis-bridge [Aug 18 10:34:27] DEBUG[13678] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c78074930(Recorder/ARI-0000001c;2) is leaving softmix technology [Aug 18 10:34:27] DEBUG[14047] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[13754] stasis.c: Destroying topic. name: cache:199/channel:1629282838.167, detail: [Aug 18 10:34:27] DEBUG[13754] stasis.c: Topic 'cache:199/channel:1629282838.167': 0x7f0c78075540 destroyed [Aug 18 10:34:27] DEBUG[13754] stasis.c: Destroying topic. name: channel:1629282838.167, detail: [Aug 18 10:34:27] DEBUG[13754] stasis.c: Topic 'channel:1629282838.167': 0x7f0c78074b20 destroyed [Aug 18 10:34:27] DEBUG[13340] channel.c: Channel 0x7f0ca0046fc0 'Announcer/ARI-0000000b;2' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[15017] bridge_native_rtp.c: Bridge '32884ae9-cc8b-4356-9f35-b03521fb3a1e' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[14745] res_rtp_asterisk.c: (0x7f0c880b67b0) RTP audio difference is 640, ms is 60 [Aug 18 10:34:27] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.569, detail: [Aug 18 10:34:27] DEBUG[13329] channel.c: Channel 0x7f0c88047e20 'Recorder/ARI-0000000a;2' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:27] DEBUG[15039] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15039] http.c: HTTP Request URI is /ari/playbacks/74a7216f-f71d-40f9-8ec0-a1e6c6e24128 [Aug 18 10:34:27] DEBUG[15039] http.c: match request [ari/playbacks/74a7216f-f71d-40f9-8ec0-a1e6c6e24128] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15039] http.c: match request [ari/playbacks/74a7216f-f71d-40f9-8ec0-a1e6c6e24128] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15039] http.c: match request [ari/playbacks/74a7216f-f71d-40f9-8ec0-a1e6c6e24128] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15039] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Finding handler for playbacks/74a7216f-f71d-40f9-8ec0-a1e6c6e24128 [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Finding handler for playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Finding handler for 74a7216f-f71d-40f9-8ec0-a1e6c6e24128 [Aug 18 10:34:27] DEBUG[15039] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15039] res_ari.c: No explicit handler found for 74a7216f-f71d-40f9-8ec0-a1e6c6e24128. Using wildcard playbackId. [Aug 18 10:34:27] DEBUG[15039] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] DEBUG[15039] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15022] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[15022] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15017] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15017] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15017] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[15017] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15017] bridge.c: Bridge 32884ae9-cc8b-4356-9f35-b03521fb3a1e: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15017] bridge.c: Bridge 32884ae9-cc8b-4356-9f35-b03521fb3a1e: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[15017] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[15038] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15038] http.c: HTTP Request URI is /ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/addChannel?channel=213028 [Aug 18 10:34:27] DEBUG[15038] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15038] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15038] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/addChannel] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15038] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Finding handler for bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/addChannel [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Finding handler for 413b28bc-b121-462c-8ad3-b989ef736d5a [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15038] res_ari.c: No explicit handler found for 413b28bc-b121-462c-8ad3-b989ef736d5a. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Finding handler for addChannel [Aug 18 10:34:27] DEBUG[15038] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:27] DEBUG[15038] stasis/control.c: 213028: Sending channel add_to_bridge command [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.569': 0x7f0c30070890 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:654/channel:1629282867.569, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:654/channel:1629282867.569': 0x7f0c301139c0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:654/channel:1629282867.569, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:654/channel:1629282867.569': 0x7f0c301139c0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.569, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.569': 0x7f0c30070890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a8', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213132', '')] [Aug 18 10:34:27] DEBUG[15017] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:27] DEBUG[13346] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15041] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[13799] res_rtp_asterisk.c: (0x7f0c1c0b2b20) RTP audio difference is 1104, ms is 89 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213132': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213132' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:418/channel:213132, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:418/channel:213132': 0x2c2fa80 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213132, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213132': 0x2c2cb60 destroyed [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213126': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213126' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:415/channel:213126, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:415/channel:213126': 0x7f0ca81188b0 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213126, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213126': 0x7f0ca81095c0 destroyed [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213125': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213125' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:417/channel:213125, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:417/channel:213125': 0x7f0c88055060 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213125, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213125': 0x7f0c88052930 destroyed [Aug 18 10:34:27] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15040] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15040] http.c: HTTP Request URI is /ari/channels/robot_212981 [Aug 18 10:34:27] DEBUG[15044] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15041] http.c: HTTP Request URI is /ari/channels/213027/snoop?app=calls_0&spy=in [Aug 18 10:34:27] DEBUG[15041] http.c: match request [ari/channels/213027/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15040] http.c: match request [ari/channels/robot_212981] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[13796] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[13678] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52'. Checking compatability for channels 'Announcer/ARI-00000038;2' and 'SIP/zvonobot-00000030' [Aug 18 10:34:27] DEBUG[15044] http.c: HTTP Request URI is /ari/channels/213125 [Aug 18 10:34:27] DEBUG[13678] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as channel 'SIP/zvonobot-00000030' has features which prevent it [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[13346] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:27] DEBUG[13346] channel.c: Channel Announcer/ARI-0000000b;1 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15040] http.c: match request [ari/channels/robot_212981] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15042] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.570, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.570': 0x7f0c30070890 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:655/channel:1629282867.570, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:655/channel:1629282867.570': 0x7f0c301139c0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:655/channel:1629282867.570, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:655/channel:1629282867.570': 0x7f0c301139c0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.570, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.570': 0x7f0c30070890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a5', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213126', '')] [Aug 18 10:34:27] DEBUG[15040] http.c: match request [ari/channels/robot_212981] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15040] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Finding handler for channels/robot_212981 [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Finding handler for robot_212981 [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking channels create: Didn't match robot_212981 [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15040] res_ari.c: Checking channels externalMedia: Didn't match robot_212981 [Aug 18 10:34:27] DEBUG[15040] res_ari.c: No explicit handler found for robot_212981. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[13678] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15041] http.c: match request [ari/channels/213027/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:27] NOTICE[13346] res_stasis_playback.c: 1629282833.94: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0;received=159.65.48.104 From: ;tag=as66ca64f7 To: ;tag=as64d137ce Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75ad3fff" Content-Length: 0 <-------------> [Aug 18 10:34:27] VERBOSE[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: switching from softmix technology to simple_bridge [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03786bb0;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as66ca64f7 [Aug 18 10:34:27] DEBUG[13627] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15041] http.c: match request [ari/channels/213027/snoop] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:27] DEBUG[15044] http.c: match request [ari/channels/213125] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[13627] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15042] http.c: HTTP Request URI is /ari/channels/213132 [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0c24057ae0(Announcer/ARI-00000038;2) to dummy bridge temporarily [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: moving 0x7f0ca0073e00(SIP/zvonobot-00000030) to dummy bridge temporarily [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c24057ae0(Announcer/ARI-00000038;2) is leaving softmix technology (dummy) [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is leaving softmix technology (dummy) [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology stop [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c24057ae0(Announcer/ARI-00000038;2) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[13678] channel.c: Channel Announcer/ARI-00000038;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[13678] channel.c: Channel Announcer/ARI-00000038;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[13678] channel.c: Channel Announcer/ARI-00000038;2 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[13678] channel.c: Channel Announcer/ARI-00000038;2 setting read format path: slin -> slin [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: deferring softmix technology destructor [Aug 18 10:34:27] DEBUG[13678] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: queueing action type:13 sub:1000 [Aug 18 10:34:27] DEBUG[15041] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as64d137ce [Aug 18 10:34:27] DEBUG[15043] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:27] DEBUG[13627] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15043] http.c: HTTP Request URI is /ari/channels/213126 [Aug 18 10:34:27] DEBUG[15043] http.c: match request [ari/channels/213126] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15043] http.c: match request [ari/channels/213126] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15043] http.c: match request [ari/channels/213126] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15043] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Finding handler for channels/213126 [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Finding handler for 213126 [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking channels create: Didn't match 213126 [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15043] res_ari.c: Checking channels externalMedia: Didn't match 213126 [Aug 18 10:34:27] DEBUG[15043] res_ari.c: No explicit handler found for 213126. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[13627] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75ad3fff" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 (Checking To) --From tag as66ca64f7 --To-tag as64d137ce [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (3) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (3) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[13627] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[13627] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee is already using the new technology. [Aug 18 10:34:27] DEBUG[15044] http.c: match request [ari/channels/213125] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Finding handler for channels/213027/snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Finding handler for 213027 [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channels create: Didn't match 213027 [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channels externalMedia: Didn't match 213027 [Aug 18 10:34:27] DEBUG[15041] res_ari.c: No explicit handler found for 213027. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Finding handler for snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:27] DEBUG[15041] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:27] DEBUG[13346] channel.c: Channel 0x7f0ca003f150 'Announcer/ARI-0000000b;1' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[15042] http.c: match request [ari/channels/213132] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15044] http.c: match request [ari/channels/213125] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x2c86070) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x2c86070) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x2c86070) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[13679] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: pulling 0x7f0c94055bb0(Recorder/ARI-0000001e;2) [Aug 18 10:34:27] VERBOSE[13679] bridge_channel.c: Channel Recorder/ARI-0000001e;2 left 'simple_bridge' stasis-bridge <357a4882-a24d-489f-8ff8-98badd81b2ee> [Aug 18 10:34:27] DEBUG[13679] bridge_channel.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: 0x7f0c94055bb0(Recorder/ARI-0000001e;2) is leaving simple_bridge technology [Aug 18 10:34:27] DEBUG[15042] http.c: match request [ari/channels/213132] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15044] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14857] channel.c: Channel 0x7f0ca4008f00 'SIP/zvonobot-000000d2' allocated [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:27] DEBUG[13679] bridge_native_rtp.c: Bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:27] DEBUG[13679] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x2c86070) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x2c86070' [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[13679] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2f470efb6aa14c6e01e982ef5bf49a87@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[13679] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[13679] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[13679] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee is already using the new technology. [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:27] DEBUG[14857] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:27] DEBUG[13679] channel.c: Channel 0x7f0c9406b7d0 'Recorder/ARI-0000001e;2' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88038c40) DTLS stop [Aug 18 10:34:27] DEBUG[15042] http.c: match request [ari/channels/213132] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Bridge is returning 0x7f0ca803dbf0(SIP/zvonobot-0000003b) to read format alaw [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.571, detail: [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Finding handler for channels/213125 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88038c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88038c40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c88038c40) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c88038c40' [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7f79f66c036e43a76469b60e207fd23d@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca80ebb80) DTLS stop [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca80ebb80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca80ebb80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca80ebb80) ICE RTP transport deallocating [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca80ebb80' [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6328ms with no response [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Hanging up call 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.571': 0x7f0c30070890 created [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[15042] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[14741] channel.c: Channel 0x7f0ca811b900 'SIP/zvonobot-000000b6' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:656/channel:1629282867.571, detail: [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:656/channel:1629282867.571': 0x7f0c301139c0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:656/channel:1629282867.571, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:656/channel:1629282867.571': 0x7f0c301139c0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.571, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.571': 0x7f0c30070890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a7', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213125', '')] [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Finding handler for channels/213132 [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Finding handler for 213132 [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking channels create: Didn't match 213132 [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15042] res_ari.c: Checking channels externalMedia: Didn't match 213132 [Aug 18 10:34:27] DEBUG[15042] res_ari.c: No explicit handler found for 213132. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[13627] channel.c: Channel SIP/zvonobot-0000003b setting read format path: ulaw -> alaw [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[13627] bridge_channel.c: Bridge is returning 0x7f0ca803dbf0(SIP/zvonobot-0000003b) to write format alaw [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (4) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[14857] res_stasis.c: calls_0: Subscribing to 213176 [Aug 18 10:34:27] DEBUG[14857] stasis/app.c: Channel '213176' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[13619] channel.c: SIP/zvonobot-00000030: Dropping redundant connected line update "" <>. [Aug 18 10:34:27] DEBUG[20534] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:27] DEBUG[14730] bridge_softmix.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: stopping mixing thread [Aug 18 10:34:27] DEBUG[13619] res_rtp_asterisk.c: (0x7f0c9801aec0) RTP audio difference is 1072, ms is 154 [Aug 18 10:34:27] DEBUG[20534] bridge_softmix.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: Waiting for mixing thread to die. [Aug 18 10:34:27] DEBUG[14857] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[13678] channel.c: Channel 0x7f0c78059c80 'Recorder/ARI-0000001c;2' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[13627] channel.c: Channel SIP/zvonobot-0000003b setting write format path: alaw -> ulaw [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[14857] http.c: HTTP closing session. Top level [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1995206f;received=159.65.48.104 From: ;tag=as5c6f3360 To: ;tag=as39f5e28d Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 821608543 821608543 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16216 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[14588] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:27] DEBUG[14588] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Outgoing Call for 79821116864 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1995206f;received=159.65.48.104 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5c6f3360 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as39f5e28d [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-0000003b - start 1629282832.357853 answer 1629282838.223954 end 1629282867.639527 dur 35.281 bill 29.415 dispo ANSWERED [Aug 18 10:34:27] DEBUG[13627] stasis/control.c: 213023, 357a4882-a24d-489f-8ff8-98badd81b2ee: Channel was departed from bridge [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Finding handler for 213125 [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking channels create: Didn't match 213125 [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15044] res_ari.c: Checking channels externalMedia: Didn't match 213125 [Aug 18 10:34:27] DEBUG[15044] res_ari.c: No explicit handler found for 213125. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[13627] stasis/app.c: bridge '357a4882-a24d-489f-8ff8-98badd81b2ee': is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[13627] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:27] DEBUG[13292] stasis/control.c: 213023: Channel departing bridge [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:27] DEBUG[13292] bridge.c: Waiting for 0x7f0ca803dbf0(SIP/zvonobot-0000003b) bridge thread to die. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:27] DEBUG[13292] stasis/app.c: channel '213023': is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[14890] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:27] DEBUG[14890] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:27] DEBUG[14890] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:27] DEBUG[14890] channel.c: Channel Announcer/ARI-0000003f;1 setting write format path: slin -> slin [Aug 18 10:34:27] DEBUG[14890] channel.c: Channel 0x7f0c940adc90 'Announcer/ARI-0000003f;1' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:27] DEBUG[13292] channel.c: Channel 0x7f0c7c03c750 'SIP/zvonobot-0000003b' hanging up. Refs: 3 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:27] DEBUG[13426] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000044 [Aug 18 10:34:27] DEBUG[13426] stasis/control.c: 213028: Adding to bridge 413b28bc-b121-462c-8ad3-b989ef736d5a [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b6 - start 1629282861.240960 answer 0.000000 end 1629282867.733813 dur 6.492 bill 1629282867.733 dispo NO ANSWER [Aug 18 10:34:27] DEBUG[13426] stasis/app.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 821608543 821608543 IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16216 RTP/AVP 0 8 101 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:27] DEBUG[15048] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: 0x7f0ca803dbf0(SIP/zvonobot-00000044) is joining [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 (Checking To) --From tag as5c6f3360 --To-tag as39f5e28d [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:27] VERBOSE[15045] chan_sip.c: Audio is at 16258 [Aug 18 10:34:27] VERBOSE[15045] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:27] VERBOSE[15045] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:27] VERBOSE[15045] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Initializing initreq for method INVITE - callid 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116864@178.62.121.41 SIP/2.0 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 3 [ 52]: From: ;tag=as220fc8e1 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 6 [ 60]: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Stopping retransmission on '543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:27 GMT [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Got SDP version 821608543 and unique parts [root 821608543 IN IP4 178.62.121.41] [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 821608543 821608543 IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:27] VERBOSE[15045] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:27] DEBUG[15048] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: pushing 0x7f0ca803dbf0(SIP/zvonobot-00000044) [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:27] DEBUG[15045] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:27] VERBOSE[15048] bridge_channel.c: Channel SIP/zvonobot-00000044 joined 'simple_bridge' stasis-bridge <413b28bc-b121-462c-8ad3-b989ef736d5a> [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:27] VERBOSE[15045] dial.c: Called zvonobot/79821116864 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2003ba40) ICE set role failed; no ice instance [Aug 18 10:34:27] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2003ba40) RTCP setting address on RTP instance [Aug 18 10:34:27] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c20044340 -- Strict RTP learning after remote address set to: 178.62.121.41:16216 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:16216 [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00975c8) from 0x7f0c147e2330 to 0x7f0c2003bc18 [Aug 18 10:34:27] DEBUG[15048] bridge_native_rtp.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0076dd8) from 0x7f0c147e2330 to 0x7f0c2003bc18 [Aug 18 10:34:27] DEBUG[15048] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15048] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15048] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15048] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15048] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a is already using the new technology. [Aug 18 10:34:27] DEBUG[15048] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: 0x7f0ca803dbf0(SIP/zvonobot-00000044) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb00e4458) from 0x7f0c147e2330 to 0x7f0c2003bc18 [Aug 18 10:34:27] DEBUG[13426] stasis/app.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2003ba40) RTCP ignoring duplicate property [Aug 18 10:34:27] DEBUG[15048] res_rtp_asterisk.c: (0x7f0c3c031e30) RTP changing ssrc from 1382101575 to 1149007546 due to a source change [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000040 setting read format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000040 setting write format path: alaw -> alaw [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2003ba40) DTLS - ast_rtp_activate rtp=0x7f0c20044340 - setup and perform DTLS' [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20044340) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20044340) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:27] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Strict routing enforced for session 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117011@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK03c7950f Max-Forwards: 70 From: ;tag=as5c6f3360 To: ;tag=as39f5e28d Contact: Call-ID: 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6412ms with no response [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Hanging up call 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #100 (5) INVITE - 5 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #100)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336558 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (4) BYE - 8 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6135ms with no response [Aug 18 10:34:27] WARNING[20585] chan_sip.c: Hanging up call 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Session timer started: 65 - 543ad357098faf3508a4b3cb38cdf0ca@159.65.48.104:5060 1768000ms [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK3d6b815d Max-Forwards: 70 From: ;tag=as2a9101f2 To: ;tag=as28b45d6b Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="0a66dcb3", response="012999b0561cf01858c8b3c0fe804b3e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK3d6b815d [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as2a9101f2 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as28b45d6b [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="0a66dcb3", response="012999b0561cf01858c8b3c0fe804b3e" [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:27] DEBUG[15038] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] VERBOSE[13419] dial.c: SIP/zvonobot-00000040 answered [Aug 18 10:34:27] VERBOSE[13419] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000040 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:27] DEBUG[15038] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[14753] channel.c: Channel 0x7f0c9808a1b0 'SIP/zvonobot-000000b9' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[14744] channel.c: Channel 0x7f0cb0169950 'SIP/zvonobot-000000b7' hanging up. Refs: 2 [Aug 18 10:34:27] DEBUG[13419] stasis/app.c: Channel '213029' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[15052] http.c: HTTP opening session. Top level [Aug 18 10:34:27] VERBOSE[13419] res_rtp_asterisk.c: 0x7f0c20044340 -- Strict RTP switching to RTP target address 178.62.121.41:16216 as source [Aug 18 10:34:27] DEBUG[13419] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:27] DEBUG[13419] channel.c: Channel SIP/zvonobot-00000040 setting read format path: ulaw -> alaw [Aug 18 10:34:27] DEBUG[13419] channel.c: Channel SIP/zvonobot-00000040 setting write format path: alaw -> ulaw [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:27] DEBUG[15052] http.c: HTTP Request URI is /ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:27] DEBUG[15052] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/play] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[14851] channel.c: Channel 0x7f0c9c028e70 'SIP/zvonobot-000000d3' allocated [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:27] DEBUG[14851] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:27] DEBUG[14851] res_stasis.c: calls_0: Subscribing to 213177 [Aug 18 10:34:27] DEBUG[14851] stasis/app.c: Channel '213177' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Outgoing Call for 79821116863 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:27] VERBOSE[15053] chan_sip.c: Audio is at 10056 [Aug 18 10:34:27] VERBOSE[15053] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b7 - start 1629282861.412514 answer 0.000000 end 1629282867.887875 dur 6.475 bill 1629282867.887 dispo NO ANSWER [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 (Checking From) --From tag as2a9101f2 --To-tag as28b45d6b [Aug 18 10:34:27] DEBUG[14851] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[15052] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/play] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15052] http.c: match request [ari/bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/play] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[14585] chan_sip.c: Hangup call SIP/zvonobot-00000096, SIP callid 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[14585] res_rtp_asterisk.c: (0x7f0c240f0400) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14585] res_rtp_asterisk.c: (0x7f0c240f0400) DTLS srtp - stopped timeout timer' [Aug 18 10:34:27] DEBUG[14585] channel.c: Channel 0x7f0c2413d0a0 'SIP/zvonobot-00000096' destroying [Aug 18 10:34:27] DEBUG[14851] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15052] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Finding handler for bridges/9cefb3ad-33ea-4a52-96a1-42b677d6802c/play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[13619] audiohook.c: Audiohook 0x7f0c2c0b4640 has stale audio in its factories. Flushing them both [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15051] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b9 - start 1629282861.465030 answer 0.000000 end 1629282867.889304 dur 6.424 bill 1629282867.889 dispo NO ANSWER [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Finding handler for 9cefb3ad-33ea-4a52-96a1-42b677d6802c [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15052] res_ari.c: No explicit handler found for 9cefb3ad-33ea-4a52-96a1-42b677d6802c. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Finding handler for play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:27] DEBUG[15052] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:27] DEBUG[15051] http.c: HTTP Request URI is /ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/record?name=213028_guFLOBpoQdRZKIhmRxjdjWZQKKcAqmNk&format=wav [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282867.572, detail: [Aug 18 10:34:27] VERBOSE[15053] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.572': 0x7f0c30070890 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Creating topic. name: cache:657/channel:1629282867.572, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:657/channel:1629282867.572': 0x7f0c301078e0 created [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: cache:657/channel:1629282867.572, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'cache:657/channel:1629282867.572': 0x7f0c301078e0 destroyed [Aug 18 10:34:27] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282867.572, detail: [Aug 18 10:34:27] DEBUG[20545] stasis.c: Topic 'channel:1629282867.572': 0x7f0c30070890 destroyed [Aug 18 10:34:27] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000096', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213114', '')] [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:27] DEBUG[15054] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15054] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:27] VERBOSE[15053] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:27] DEBUG[15054] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15054] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15054] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15054] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Initializing initreq for method INVITE - callid 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116863@178.62.121.41 SIP/2.0 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 [Aug 18 10:34:27] DEBUG[15054] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 3 [ 52]: From: ;tag=as48cc2656 [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213114': is 0 interested in calls_0 [Aug 18 10:34:27] DEBUG[15054] stasis.c: Creating topic. name: bridge:466c6117-1cbd-4c34-863b-5d0db95ca0e0, detail: [Aug 18 10:34:27] DEBUG[20620] stasis/app.c: channel '213114' unsubscribed from calls_0 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:27] DEBUG[15054] stasis.c: Topic 'bridge:466c6117-1cbd-4c34-863b-5d0db95ca0e0': 0x7f0c300a8da0 created [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 6 [ 60]: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15055] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15054] stasis.c: Creating topic. name: cache:658/bridge:466c6117-1cbd-4c34-863b-5d0db95ca0e0, detail: [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:27] DEBUG[15055] http.c: HTTP Request URI is /ari/channels/213114 [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: cache:376/channel:213114, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'cache:376/channel:213114': 0x7f0c2413fcb0 destroyed [Aug 18 10:34:27] DEBUG[20620] stasis.c: Destroying topic. name: channel:213114, detail: [Aug 18 10:34:27] DEBUG[20620] stasis.c: Topic 'channel:213114': 0x7f0c2413ee20 destroyed [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:27 GMT [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:27] VERBOSE[15053] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Aug 18 10:34:27] DEBUG[15053] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:27] DEBUG[15055] http.c: match request [ari/channels/213114] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15054] stasis.c: Topic 'cache:658/bridge:466c6117-1cbd-4c34-863b-5d0db95ca0e0': 0x7f0c300a8e50 created [Aug 18 10:34:27] DEBUG[15051] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/record] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15055] http.c: match request [ari/channels/213114] with handler [phoneprov] len 9 [Aug 18 10:34:27] VERBOSE[15053] dial.c: Called zvonobot/79821116863 [Aug 18 10:34:27] DEBUG[15055] http.c: match request [ari/channels/213114] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15051] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/record] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:27] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:27] DEBUG[15054] bridge_native_rtp.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15054] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15054] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15054] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:27] DEBUG[15054] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15054] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: calling simple_bridge technology constructor [Aug 18 10:34:27] DEBUG[15054] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: calling simple_bridge technology start [Aug 18 10:34:27] DEBUG[15055] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:27] DEBUG[15054] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Finding handler for channels/213114 [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Finding handler for channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Finding handler for 213114 [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking channels create: Didn't match 213114 [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15055] res_ari.c: Checking channels externalMedia: Didn't match 213114 [Aug 18 10:34:27] DEBUG[15055] res_ari.c: No explicit handler found for 213114. Using wildcard channelId. [Aug 18 10:34:27] DEBUG[15054] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15057] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15051] http.c: match request [ari/bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/record] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15057] http.c: HTTP Request URI is /ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/addChannel?channel=213029 [Aug 18 10:34:27] DEBUG[15057] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15057] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15057] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/addChannel] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15057] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Finding handler for bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/addChannel [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Finding handler for 466c6117-1cbd-4c34-863b-5d0db95ca0e0 [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15051] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15057] res_ari.c: No explicit handler found for 466c6117-1cbd-4c34-863b-5d0db95ca0e0. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Finding handler for addChannel [Aug 18 10:34:27] DEBUG[15057] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:27] DEBUG[15057] stasis/control.c: 213029: Sending channel add_to_bridge command [Aug 18 10:34:27] DEBUG[13419] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-00000040 [Aug 18 10:34:27] DEBUG[13419] stasis/control.c: 213029: Adding to bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0 [Aug 18 10:34:27] DEBUG[13419] stasis/app.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0' is 1 interested in calls_0 [Aug 18 10:34:27] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:27] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:27] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:27] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:27] DEBUG[15058] bridge_channel.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: 0x7f0c8c1113d0(SIP/zvonobot-00000040) is joining [Aug 18 10:34:27] DEBUG[15058] bridge_channel.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: pushing 0x7f0c8c1113d0(SIP/zvonobot-00000040) [Aug 18 10:34:27] VERBOSE[15058] bridge_channel.c: Channel SIP/zvonobot-00000040 joined 'simple_bridge' stasis-bridge <466c6117-1cbd-4c34-863b-5d0db95ca0e0> [Aug 18 10:34:27] DEBUG[15058] bridge_native_rtp.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0' can not use native RTP bridge as two channels are required [Aug 18 10:34:27] DEBUG[15058] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:27] DEBUG[15058] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Finding handler for bridges/413b28bc-b121-462c-8ad3-b989ef736d5a/record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15058] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:27] DEBUG[15058] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15058] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0 is already using the new technology. [Aug 18 10:34:27] DEBUG[15058] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: 0x7f0c8c1113d0(SIP/zvonobot-00000040) is joining simple_bridge technology [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15058] res_rtp_asterisk.c: (0x7f0c2003ba40) RTP changing ssrc from 2128633843 to 1934803986 due to a source change [Aug 18 10:34:27] DEBUG[13419] stasis/app.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0' is 2 interested in calls_0 [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15057] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Finding handler for 413b28bc-b121-462c-8ad3-b989ef736d5a [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15051] res_ari.c: No explicit handler found for 413b28bc-b121-462c-8ad3-b989ef736d5a. Using wildcard bridgeId. [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Finding handler for record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:27] DEBUG[15057] http.c: HTTP closing session. Top level [Aug 18 10:34:27] DEBUG[15059] http.c: HTTP opening session. Top level [Aug 18 10:34:27] DEBUG[15059] http.c: HTTP Request URI is /ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/record?name=213029_wGlDyvoiSvQdgQwKfOVYDJUhnGMUAKBe&format=wav [Aug 18 10:34:27] DEBUG[15051] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:27] DEBUG[15059] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/record] with handler [httpstatus] len 10 [Aug 18 10:34:27] DEBUG[15059] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/record] with handler [phoneprov] len 9 [Aug 18 10:34:27] DEBUG[15059] http.c: match request [ari/bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/record] with handler [ari] len 3 [Aug 18 10:34:27] DEBUG[15059] http.c: Match made with [ari] [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Finding handler for bridges/466c6117-1cbd-4c34-863b-5d0db95ca0e0/record [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Finding handler for bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:27] DEBUG[15051] stasis.c: Creating topic. name: channel:1629282867.573, detail: [Aug 18 10:34:27] DEBUG[15051] stasis.c: Topic 'channel:1629282867.573': 0x7f0c2c06ac80 created [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Finding handler for 466c6117-1cbd-4c34-863b-5d0db95ca0e0 [Aug 18 10:34:27] DEBUG[15051] stasis.c: Creating topic. name: cache:659/channel:1629282867.573, detail: [Aug 18 10:34:27] DEBUG[15059] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:27] DEBUG[15059] res_ari.c: No explicit handler found for 466c6117-1cbd-4c34-863b-5d0db95ca0e0. Using wildcard bridgeId. [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Finding handler for record [Aug 18 10:34:27] DEBUG[15051] stasis.c: Topic 'cache:659/channel:1629282867.573': 0x7f0c2c0ad500 created [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:28] DEBUG[15059] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:28] DEBUG[15059] stasis.c: Creating topic. name: channel:1629282867.574, detail: [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[15059] stasis.c: Topic 'channel:1629282867.574': 0x7f0c74021d00 created [Aug 18 10:34:28] DEBUG[15059] stasis.c: Creating topic. name: cache:660/channel:1629282867.574, detail: [Aug 18 10:34:28] DEBUG[14865] channel.c: Channel 0x7f0cb40a3910 'SIP/zvonobot-000000d4' allocated [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[15059] stasis.c: Topic 'cache:660/channel:1629282867.574': 0x7f0c74021e50 created [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Setting 0x7f0c20083ee0(SIP/zvonobot-00000014) state from:0 to:1 [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14865] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: pulling 0x7f0c20083ee0(SIP/zvonobot-00000014) [Aug 18 10:34:28] VERBOSE[14637] bridge_channel.c: Channel SIP/zvonobot-00000014 left 'simple_bridge' stasis-bridge [Aug 18 10:34:28] DEBUG[14865] res_stasis.c: calls_0: Subscribing to 213179 [Aug 18 10:34:28] DEBUG[14865] stasis/app.c: Channel '213179' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0c20083ee0(SIP/zvonobot-00000014) is leaving simple_bridge technology [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Setting 0x7f0cac01a9f0(Recorder/ARI-00000043;2) state from:0 to:2 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:28] DEBUG[14865] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14865] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK3d6b815d;received=178.62.121.41 From: ;tag=as2a9101f2 To: ;tag=as28b45d6b Call-ID: 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14637] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[14637] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[14637] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000014 - start 1629282826.005291 answer 1629282858.807157 end 1629282868.023503 dur 42.018 bill 9.216 dispo ANSWERED [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6388ms with no response [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Hanging up call 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:28] DEBUG[14637] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14637] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14637] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10 is already using the new technology. [Aug 18 10:34:28] DEBUG[14941] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: pulling 0x7f0cac01a9f0(Recorder/ARI-00000043;2) [Aug 18 10:34:28] VERBOSE[14941] bridge_channel.c: Channel Recorder/ARI-00000043;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:28] DEBUG[14941] bridge_channel.c: Bridge f495d952-07a0-4425-9378-2616afbaca10: 0x7f0cac01a9f0(Recorder/ARI-00000043;2) is leaving simple_bridge technology [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Bridge is returning 0x7f0c20083ee0(SIP/zvonobot-00000014) to read format alaw [Aug 18 10:34:28] DEBUG[14637] channel.c: Channel SIP/zvonobot-00000014 setting read format path: alaw -> alaw [Aug 18 10:34:28] DEBUG[14637] bridge_channel.c: Bridge is returning 0x7f0c20083ee0(SIP/zvonobot-00000014) to write format alaw [Aug 18 10:34:28] DEBUG[14637] channel.c: Channel SIP/zvonobot-00000014 setting write format path: alaw -> alaw [Aug 18 10:34:28] DEBUG[14637] stasis/control.c: 212984, f495d952-07a0-4425-9378-2616afbaca10: Channel was departed from bridge [Aug 18 10:34:28] DEBUG[14637] stasis/app.c: bridge 'f495d952-07a0-4425-9378-2616afbaca10': is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14637] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:28] DEBUG[12982] stasis/control.c: 212984: Channel departing bridge [Aug 18 10:34:28] DEBUG[12982] bridge.c: Waiting for 0x7f0c20083ee0(SIP/zvonobot-00000014) bridge thread to die. [Aug 18 10:34:28] DEBUG[14941] bridge_native_rtp.c: Bridge 'f495d952-07a0-4425-9378-2616afbaca10' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[14941] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[14941] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14941] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[12982] stasis/app.c: channel '212984': is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14941] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Outgoing Call for 79821116861 [Aug 18 10:34:28] DEBUG[14941] bridge.c: Bridge f495d952-07a0-4425-9378-2616afbaca10 is already using the new technology. [Aug 18 10:34:28] DEBUG[14752] channel.c: Channel 0x7f0c08046230 'SIP/zvonobot-000000b8' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[12982] channel.c: Channel 0x7f0c24021660 'SIP/zvonobot-00000014' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000b8 - start 1629282861.526413 answer 0.000000 end 1629282868.032513 dur 6.506 bill 1629282868.032 dispo NO ANSWER [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:28] DEBUG[14941] channel.c: Channel 0x7f0cac05f500 'Recorder/ARI-00000043;2' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[15062] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15062] http.c: HTTP Request URI is /ari/channels/213194?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116846&callerId=74950493843 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:28] DEBUG[15062] http.c: match request [ari/channels/213194] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[14536] iostream.c: TCP socket error reading data: Connection reset by peer [Aug 18 10:34:28] DEBUG[15062] http.c: match request [ari/channels/213194] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[14536] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15062] http.c: match request [ari/channels/213194] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15062] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[15062] http.c: HTTP consuming request body [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Finding handler for channels/213194 [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15064] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15064] http.c: HTTP Request URI is /ari/channels/212969/snoop?app=calls_0&spy=in [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Finding handler for 213194 [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking channels create: Didn't match 213194 [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15062] res_ari.c: Checking channels externalMedia: Didn't match 213194 [Aug 18 10:34:28] DEBUG[15062] res_ari.c: No explicit handler found for 213194. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Session timer stopped: 83 - 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15064] http.c: match request [ari/channels/212969/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] VERBOSE[15060] chan_sip.c: Audio is at 18760 [Aug 18 10:34:28] DEBUG[15064] http.c: match request [ari/channels/212969/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:28] VERBOSE[15060] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] DEBUG[15064] http.c: match request [ari/channels/212969/snoop] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15064] http.c: Match made with [ari] [Aug 18 10:34:28] VERBOSE[15060] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Finding handler for channels/212969/snoop [Aug 18 10:34:28] VERBOSE[15060] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15065] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Finding handler for 212969 [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channels create: Didn't match 212969 [Aug 18 10:34:28] DEBUG[15065] http.c: HTTP Request URI is /ari/channels/213195?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116845&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channels externalMedia: Didn't match 212969 [Aug 18 10:34:28] DEBUG[15064] res_ari.c: No explicit handler found for 212969. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Finding handler for snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:28] DEBUG[15065] http.c: match request [ari/channels/213195] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15064] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:28] DEBUG[15065] http.c: match request [ari/channels/213195] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15065] http.c: match request [ari/channels/213195] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15065] http.c: Match made with [ari] [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732;received=159.65.48.104 From: ;tag=as6a49d994 To: ;tag=as0fc9f26c Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11f4d062" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[15065] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Finding handler for channels/213195 [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Initializing initreq for method INVITE - callid 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6a49d994 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0fc9f26c [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Finding handler for 213195 [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking channels create: Didn't match 213195 [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15065] res_ari.c: Checking channels externalMedia: Didn't match 213195 [Aug 18 10:34:28] DEBUG[15065] res_ari.c: No explicit handler found for 213195. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116861@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15068] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: Allocating new SIP dialog for 449742ca460f159e22deaffe0d40376e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11f4d062" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 (Checking To) --From tag as6a49d994 --To-tag as0fc9f26c [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Stopping retransmission on '29b2374b344692165b25e9de23da23d4@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116899@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1f763732 Max-Forwards: 70 From: ;tag=as6a49d994 To: Contact: Call-ID: 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6263ms with no response [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Hanging up call 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000ba - start 1629282861.550834 answer 0.000000 end 1629282868.124724 dur 6.573 bill 1629282868.124 dispo NO ANSWER [Aug 18 10:34:28] DEBUG[15062] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c053b50' [Aug 18 10:34:28] DEBUG[14754] channel.c: Channel 0x7f0c200cad60 'SIP/zvonobot-000000ba' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) RTP allocated port 19816 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:28] DEBUG[15068] http.c: HTTP Request URI is /ari/channels/213196?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116844&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15068] http.c: match request [ari/channels/213196] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15068] http.c: match request [ari/channels/213196] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15068] http.c: match request [ari/channels/213196] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15068] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15068] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Finding handler for channels/213196 [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Finding handler for 213196 [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking channels create: Didn't match 213196 [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15068] res_ari.c: Checking channels externalMedia: Didn't match 213196 [Aug 18 10:34:28] DEBUG[15068] res_ari.c: No explicit handler found for 213196. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE creating session 0.0.0.0:19816 (19816) [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE create [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15073] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE add system candidates [Aug 18 10:34:28] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:28] DEBUG[15062] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15062] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE add candidate: 159.65.48.104:19816, 2130706431 [Aug 18 10:34:28] DEBUG[15062] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15062] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15073] http.c: HTTP Request URI is /ari/channels/213201?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116839&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE add candidate: 10.131.0.10:19816, 2130706431 [Aug 18 10:34:28] DEBUG[15062] rtp_engine.c: RTP instance '0x7f0c7c053b50' is setup and ready to go [Aug 18 10:34:28] DEBUG[15073] http.c: match request [ari/channels/213201] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15076] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 [Aug 18 10:34:28] DEBUG[15073] http.c: match request [ari/channels/213201] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) ICE stopped [Aug 18 10:34:28] DEBUG[15062] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15076] http.c: HTTP Request URI is /ari/channels/213197?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116843&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15062] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Destroying SIP dialog 780fccc405c242d348e2e247384adc25@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15073] http.c: match request [ari/channels/213201] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15076] http.c: match request [ari/channels/213197] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15073] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15073] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15062] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15062] res_rtp_asterisk.c: (0x7f0c7c053b50) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15062] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15062] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15062] chan_sip.c: SIP call-id changed from '449742ca460f159e22deaffe0d40376e@127.0.1.1:5060' to '6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15062] stasis.c: Creating topic. name: channel:213194, detail: [Aug 18 10:34:28] DEBUG[15062] stasis.c: Topic 'channel:213194': 0x7f0c7c0c5f30 created [Aug 18 10:34:28] DEBUG[15062] stasis.c: Creating topic. name: cache:661/channel:213194, detail: [Aug 18 10:34:28] DEBUG[15062] stasis.c: Topic 'cache:661/channel:213194': 0x7f0c7c0adb50 created [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '780fccc405c242d348e2e247384adc25@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c240f0400) DTLS stop [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c240f0400) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c240f0400) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c240f0400) ICE RTP transport deallocating [Aug 18 10:34:28] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c240f0400' [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15076] http.c: match request [ari/channels/213197] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 3 [ 52]: From: ;tag=as29706635 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Finding handler for channels/213201 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15076] http.c: match request [ari/channels/213197] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15076] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Finding handler for 213201 [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking channels create: Didn't match 213201 [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15073] res_ari.c: Checking channels externalMedia: Didn't match 213201 [Aug 18 10:34:28] DEBUG[15073] res_ari.c: No explicit handler found for 213201. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15076] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15077] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 6 [ 60]: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15077] http.c: HTTP Request URI is /ari/channels/213199?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116841&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Finding handler for channels/213197 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f;received=159.65.48.104 From: ;tag=as056b2e4e To: ;tag=as1bdb31eb Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56b46360" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Finding handler for 213197 [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking channels create: Didn't match 213197 [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15076] res_ari.c: Checking channels externalMedia: Didn't match 213197 [Aug 18 10:34:28] DEBUG[15076] res_ari.c: No explicit handler found for 213197. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15077] http.c: match request [ari/channels/213199] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[15077] http.c: match request [ari/channels/213199] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15077] http.c: match request [ari/channels/213199] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15077] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as056b2e4e [Aug 18 10:34:28] DEBUG[15078] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: Allocating new SIP dialog for 3a2860107e20dfef430792d966aeab07@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15065] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c841259b0' [Aug 18 10:34:28] DEBUG[15077] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1bdb31eb [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56b46360" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 (Checking To) --From tag as056b2e4e --To-tag as1bdb31eb [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) RTP allocated port 19406 [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Finding handler for channels/213199 [Aug 18 10:34:28] DEBUG[15078] http.c: HTTP Request URI is /ari/channels/213200?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116840&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15079] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE creating session 0.0.0.0:19406 (19406) [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] DEBUG[15079] http.c: HTTP Request URI is /ari/channels/213202?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116838&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE create [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Finding handler for 213199 [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking channels create: Didn't match 213199 [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15077] res_ari.c: Checking channels externalMedia: Didn't match 213199 [Aug 18 10:34:28] VERBOSE[15060] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15077] res_ari.c: No explicit handler found for 213199. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE add system candidates [Aug 18 10:34:28] DEBUG[15065] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15065] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE add candidate: 159.65.48.104:19406, 2130706431 [Aug 18 10:34:28] DEBUG[15065] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15065] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE add candidate: 10.131.0.10:19406, 2130706431 [Aug 18 10:34:28] DEBUG[15065] rtp_engine.c: RTP instance '0x7f0c841259b0' is setup and ready to go [Aug 18 10:34:28] DEBUG[15078] http.c: match request [ari/channels/213200] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) ICE stopped [Aug 18 10:34:28] DEBUG[15065] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15065] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15065] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15065] res_rtp_asterisk.c: (0x7f0c841259b0) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15065] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15079] http.c: match request [ari/channels/213202] with handler [httpstatus] len 10 [Aug 18 10:34:28] WARNING[13852] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-00000024;1 [Aug 18 10:34:28] DEBUG[15079] http.c: match request [ari/channels/213202] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15065] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15078] http.c: match request [ari/channels/213200] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15078] http.c: match request [ari/channels/213200] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15065] chan_sip.c: SIP call-id changed from '3a2860107e20dfef430792d966aeab07@127.0.1.1:5060' to '27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15078] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15079] http.c: match request [ari/channels/213202] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15065] stasis.c: Creating topic. name: channel:213195, detail: [Aug 18 10:34:28] DEBUG[15078] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Finding handler for channels/213200 [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Finding handler for 213200 [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking channels create: Didn't match 213200 [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15078] res_ari.c: Checking channels externalMedia: Didn't match 213200 [Aug 18 10:34:28] DEBUG[15078] res_ari.c: No explicit handler found for 213200. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15080] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #33 [Aug 18 10:34:28] DEBUG[14931] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:28] DEBUG[14931] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:28] DEBUG[14931] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:28] DEBUG[14931] channel.c: Channel Announcer/ARI-00000042;1 setting write format path: slin -> slin [Aug 18 10:34:28] DEBUG[14931] channel.c: Channel 0x7f0c74079320 'Announcer/ARI-00000042;1' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:28] DEBUG[15080] http.c: HTTP Request URI is /ari/channels/213203?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116837&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15065] stasis.c: Topic 'channel:213195': 0x7f0c84103950 created [Aug 18 10:34:28] DEBUG[15065] stasis.c: Creating topic. name: cache:662/channel:213195, detail: [Aug 18 10:34:28] DEBUG[15065] stasis.c: Topic 'cache:662/channel:213195': 0x7f0c8408fe10 created [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Stopping retransmission on '7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:28] DEBUG[15060] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14394] res_rtp_asterisk.c: (0x7f0c38023850) RTCP got report of 100 bytes from 178.62.121.41:12659 [Aug 18 10:34:28] DEBUG[15079] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15080] http.c: match request [ari/channels/213203] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: Allocating new SIP dialog for 74ebd8eb64273f602af317363d08c26d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15073] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c14b520' [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) RTP allocated port 15154 [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE creating session 0.0.0.0:15154 (15154) [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE create [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE add system candidates [Aug 18 10:34:28] DEBUG[15073] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15073] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE add candidate: 159.65.48.104:15154, 2130706431 [Aug 18 10:34:28] DEBUG[15073] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15080] http.c: match request [ari/channels/213203] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15080] http.c: match request [ari/channels/213203] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15073] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15080] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15079] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE add candidate: 10.131.0.10:15154, 2130706431 [Aug 18 10:34:28] DEBUG[15073] rtp_engine.c: RTP instance '0x7f0c8c14b520' is setup and ready to go [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) ICE stopped [Aug 18 10:34:28] DEBUG[15073] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15073] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15073] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15073] res_rtp_asterisk.c: (0x7f0c8c14b520) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15073] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15073] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15073] chan_sip.c: SIP call-id changed from '74ebd8eb64273f602af317363d08c26d@127.0.1.1:5060' to '307304ec558a22464cac7a9262f07b46@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Finding handler for channels/213202 [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15081] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15073] stasis.c: Creating topic. name: channel:213201, detail: [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Finding handler for 213202 [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking channels create: Didn't match 213202 [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15079] res_ari.c: Checking channels externalMedia: Didn't match 213202 [Aug 18 10:34:28] DEBUG[15079] res_ari.c: No explicit handler found for 213202. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15081] http.c: HTTP Request URI is /ari/channels/213198?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116842&callerId=74950493843 [Aug 18 10:34:28] DEBUG[15080] http.c: HTTP consuming request body [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116905@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62d2e35f Max-Forwards: 70 From: ;tag=as056b2e4e To: Contact: Call-ID: 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: Allocating new SIP dialog for 44ef159a0a7505ea08e4346b38731a4a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15077] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c94091d70' [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) RTP allocated port 15450 [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE creating session 0.0.0.0:15450 (15450) [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE create [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Finding handler for channels/213203 [Aug 18 10:34:28] DEBUG[15081] http.c: match request [ari/channels/213198] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE add system candidates [Aug 18 10:34:28] DEBUG[15073] stasis.c: Topic 'channel:213201': 0x7f0c8c148770 created [Aug 18 10:34:28] DEBUG[15081] http.c: match request [ari/channels/213198] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (2) INVITE - 5 [Aug 18 10:34:28] DEBUG[15081] http.c: match request [ari/channels/213198] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15073] stasis.c: Creating topic. name: cache:663/channel:213201, detail: [Aug 18 10:34:28] DEBUG[15073] stasis.c: Topic 'cache:663/channel:213201': 0x7f0c8c06e5d0 created [Aug 18 10:34:28] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:28] DEBUG[15077] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15081] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15077] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[15081] http.c: HTTP consuming request body [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[13444] app.c: One waitfor failed, trying another [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Finding handler for channels/213198 [Aug 18 10:34:28] VERBOSE[15060] dial.c: Called zvonobot/79821116861 [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE add candidate: 159.65.48.104:15450, 2130706431 [Aug 18 10:34:28] DEBUG[15077] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15077] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: Allocating new SIP dialog for 1fd4d0286ac6a44f076e03f2662107e0@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15076] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c88080300' [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) RTP allocated port 14230 [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE creating session 0.0.0.0:14230 (14230) [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE create [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE add system candidates [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE add candidate: 10.131.0.10:15450, 2130706431 [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15076] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15076] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Finding handler for 213203 [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking channels create: Didn't match 213203 [Aug 18 10:34:28] DEBUG[14846] channel.c: Channel 0x7f0ca8066c10 'SIP/zvonobot-000000d6' allocated [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Finding handler for 213198 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #40 (5) INVITE - 5 [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #40)) [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15077] rtp_engine.c: RTP instance '0x7f0c94091d70' is setup and ready to go [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE add candidate: 159.65.48.104:14230, 2130706431 [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking channels create: Didn't match 213198 [Aug 18 10:34:28] DEBUG[15080] res_ari.c: Checking channels externalMedia: Didn't match 213203 [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14846] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204973 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15081] res_ari.c: Checking channels externalMedia: Didn't match 213198 [Aug 18 10:34:28] DEBUG[15080] res_ari.c: No explicit handler found for 213203. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15081] res_ari.c: No explicit handler found for 213198. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) ICE stopped [Aug 18 10:34:28] DEBUG[15077] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15077] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15076] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15077] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15076] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE add candidate: 10.131.0.10:14230, 2130706431 [Aug 18 10:34:28] DEBUG[15077] res_rtp_asterisk.c: (0x7f0c94091d70) RTCP setup on RTP instance [Aug 18 10:34:28] DEBUG[15076] rtp_engine.c: RTP instance '0x7f0c88080300' is setup and ready to go [Aug 18 10:34:28] DEBUG[14728] channel.c: Channel 0x7f0c080710f0 'Snoop/213036-00000018' allocated [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) ICE stopped [Aug 18 10:34:28] DEBUG[15076] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15076] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[14873] channel.c: Channel 0x2c27ac0 'SIP/zvonobot-000000d5' allocated [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14873] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[14447] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be'. Checking compatability for channels 'SIP/zvonobot-00000047' and 'Recorder/ARI-00000039;2' [Aug 18 10:34:28] DEBUG[14447] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' can not use native RTP bridge as channel 'SIP/zvonobot-00000047' has features which prevent it [Aug 18 10:34:28] DEBUG[14447] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[14447] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14447] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14447] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14447] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be is already using the new technology. [Aug 18 10:34:28] DEBUG[14728] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15076] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15076] res_rtp_asterisk.c: (0x7f0c88080300) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15076] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15076] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] VERBOSE[15077] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15076] chan_sip.c: SIP call-id changed from '1fd4d0286ac6a44f076e03f2662107e0@127.0.1.1:5060' to '136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[14728] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: Allocating new SIP dialog for 04cf135856d9cd25692d01bf5bca5edf@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6410ms with no response [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15078] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9002e460' [Aug 18 10:34:28] DEBUG[15076] stasis.c: Creating topic. name: channel:213197, detail: [Aug 18 10:34:28] DEBUG[15077] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15086] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15086] http.c: HTTP Request URI is /ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/play?media=sound%3Asilence%2F2 [Aug 18 10:34:28] DEBUG[15086] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/play] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15086] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/play] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15086] http.c: match request [ari/bridges/61075423-3ee2-4d60-8382-ee99e654a5be/play] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15086] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: Allocating new SIP dialog for 3c2b480427a0c618269f6c4555b177b0@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15079] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca811c4c0' [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) RTP allocated port 12766 [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE creating session 0.0.0.0:12766 (12766) [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE create [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE add system candidates [Aug 18 10:34:28] DEBUG[15079] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15079] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE add candidate: 159.65.48.104:12766, 2130706431 [Aug 18 10:34:28] DEBUG[15079] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15079] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE add candidate: 10.131.0.10:12766, 2130706431 [Aug 18 10:34:28] DEBUG[15079] rtp_engine.c: RTP instance '0x7f0ca811c4c0' is setup and ready to go [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) ICE stopped [Aug 18 10:34:28] DEBUG[15079] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15079] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15079] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15079] res_rtp_asterisk.c: (0x7f0ca811c4c0) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15079] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15079] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15079] chan_sip.c: SIP call-id changed from '3c2b480427a0c618269f6c4555b177b0@127.0.1.1:5060' to '104c5130341091f56623cd02618893c9@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15079] stasis.c: Creating topic. name: channel:213202, detail: [Aug 18 10:34:28] DEBUG[15089] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15077] chan_sip.c: SIP call-id changed from '44ef159a0a7505ea08e4346b38731a4a@127.0.1.1:5060' to '6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15077] stasis.c: Creating topic. name: channel:213199, detail: [Aug 18 10:34:28] DEBUG[15076] stasis.c: Topic 'channel:213197': 0x7f0c88062bc0 created [Aug 18 10:34:28] DEBUG[15076] stasis.c: Creating topic. name: cache:664/channel:213197, detail: [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) RTP allocated port 10424 [Aug 18 10:34:28] DEBUG[15089] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213036&app=calls_0&format=slin16&external_host=127.0.0.1%3A50456 [Aug 18 10:34:28] DEBUG[15079] stasis.c: Topic 'channel:213202': 0x7f0ca801a890 created [Aug 18 10:34:28] DEBUG[15079] stasis.c: Creating topic. name: cache:665/channel:213202, detail: [Aug 18 10:34:28] DEBUG[15079] stasis.c: Topic 'cache:665/channel:213202': 0x7f0ca81124b0 created [Aug 18 10:34:28] DEBUG[15076] stasis.c: Topic 'cache:664/channel:213197': 0x7f0c88040590 created [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE creating session 0.0.0.0:10424 (10424) [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE create [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE add system candidates [Aug 18 10:34:28] DEBUG[15078] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15078] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE add candidate: 159.65.48.104:10424, 2130706431 [Aug 18 10:34:28] DEBUG[15078] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15078] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE add candidate: 10.131.0.10:10424, 2130706431 [Aug 18 10:34:28] DEBUG[15078] rtp_engine.c: RTP instance '0x7f0c9002e460' is setup and ready to go [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) ICE stopped [Aug 18 10:34:28] DEBUG[15078] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15078] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15078] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15078] res_rtp_asterisk.c: (0x7f0c9002e460) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15078] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15089] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Hanging up call 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Finding handler for bridges/61075423-3ee2-4d60-8382-ee99e654a5be/play [Aug 18 10:34:28] DEBUG[15089] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15077] stasis.c: Topic 'channel:213199': 0x7f0c94015cd0 created [Aug 18 10:34:28] DEBUG[15077] stasis.c: Creating topic. name: cache:666/channel:213199, detail: [Aug 18 10:34:28] DEBUG[15077] stasis.c: Topic 'cache:666/channel:213199': 0x7f0c94039120 created [Aug 18 10:34:28] DEBUG[15089] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15089] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15078] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Finding handler for bridges [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15078] chan_sip.c: SIP call-id changed from '04cf135856d9cd25692d01bf5bca5edf@127.0.1.1:5060' to '6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15078] stasis.c: Creating topic. name: channel:213200, detail: [Aug 18 10:34:28] DEBUG[15078] stasis.c: Topic 'channel:213200': 0x7f0c9004d890 created [Aug 18 10:34:28] DEBUG[15078] stasis.c: Creating topic. name: cache:667/channel:213200, detail: [Aug 18 10:34:28] DEBUG[15078] stasis.c: Topic 'cache:667/channel:213200': 0x7f0c900806d0 created [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[14766] channel.c: Channel 0x7f0c18105090 'SIP/zvonobot-000000bc' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000bc - start 1629282861.993800 answer 0.000000 end 1629282868.516401 dur 6.522 bill 1629282868.516 dispo NO ANSWER [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:28] DEBUG[14522] channel.c: Channel 0x7f0c400b0ff0 'Recorder/ARI-00000033;2' destroying [Aug 18 10:34:28] DEBUG[14875] channel.c: Channel 0x7f0c0808d310 'SIP/zvonobot-000000d8' allocated [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14875] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[14610] chan_sip.c: Hangup call SIP/zvonobot-000000a9, SIP callid 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14610] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:28] DEBUG[14610] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[14610] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[14610] channel.c: Channel 0x7f0cb4063e00 'SIP/zvonobot-000000a9' destroying [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:28] DEBUG[14724] channel.c: Channel 0x7f0c2008f410 'Recorder/ARI-00000039;1' destroying [Aug 18 10:34:28] DEBUG[14724] stasis.c: Destroying topic. name: cache:421/channel:1629282854.365, detail: [Aug 18 10:34:28] DEBUG[14724] stasis.c: Topic 'cache:421/channel:1629282854.365': 0x7f0c200901b0 destroyed [Aug 18 10:34:28] DEBUG[14724] stasis.c: Destroying topic. name: channel:1629282854.365, detail: [Aug 18 10:34:28] DEBUG[14724] stasis.c: Topic 'channel:1629282854.365': 0x7f0c20048e90 destroyed [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (2) INVITE - 5 [Aug 18 10:34:28] DEBUG[15083] stasis/app.c: Channel '1629282864.517' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14716] bridge_channel.c: Setting 0x7f0c200663a0(Recorder/ARI-00000039;2) state from:0 to:1 [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282868.582, detail: [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'channel:1629282868.582': 0x7f0c301078e0 created [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: cache:668/channel:1629282868.582, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:668/channel:1629282868.582': 0x7f0c301437d0 created [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:28] DEBUG[20620] stasis/app.c: channel '213133': is 0 interested in calls_0 [Aug 18 10:34:28] DEBUG[20620] stasis/app.c: channel '213133' unsubscribed from calls_0 [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Finding handler for 61075423-3ee2-4d60-8382-ee99e654a5be [Aug 18 10:34:28] DEBUG[15090] http.c: HTTP opening session. Top level [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15090] http.c: HTTP Request URI is /ari/channels/213133 [Aug 18 10:34:28] DEBUG[15089] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14716] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: pulling 0x7f0c200663a0(Recorder/ARI-00000039;2) [Aug 18 10:34:28] VERBOSE[14716] bridge_channel.c: Channel Recorder/ARI-00000039;2 left 'simple_bridge' stasis-bridge <61075423-3ee2-4d60-8382-ee99e654a5be> [Aug 18 10:34:28] DEBUG[14716] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0c200663a0(Recorder/ARI-00000039;2) is leaving simple_bridge technology [Aug 18 10:34:28] DEBUG[15089] netsock2.c: Splitting '127.0.0.1:50456' into... [Aug 18 10:34:28] DEBUG[15089] netsock2.c: ...host '127.0.0.1' and port '50456'. [Aug 18 10:34:28] DEBUG[15089] netsock2.c: Splitting '127.0.0.1:50456' into... [Aug 18 10:34:28] DEBUG[15089] netsock2.c: ...host '127.0.0.1' and port '50456'. [Aug 18 10:34:28] DEBUG[15089] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:28] DEBUG[15089] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb014c030' [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) RTP allocated port 19244 [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) ICE creating session 127.0.0.1:19244 (19244) [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) ICE create [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) ICE add system candidates [Aug 18 10:34:28] DEBUG[15089] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15089] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) ICE add candidate: 159.65.48.104:19244, 2130706431 [Aug 18 10:34:28] DEBUG[15089] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15089] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15089] res_rtp_asterisk.c: (0x7f0cb014c030) ICE add candidate: 10.131.0.10:19244, 2130706431 [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: cache:668/channel:1629282868.582, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:668/channel:1629282868.582': 0x7f0c301437d0 destroyed [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282868.582, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'channel:1629282868.582': 0x7f0c301078e0 destroyed [Aug 18 10:34:28] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:18', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a9', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213133', '')] [Aug 18 10:34:28] DEBUG[15089] rtp_engine.c: RTP instance '0x7f0cb014c030' is setup and ready to go [Aug 18 10:34:28] DEBUG[15089] stasis.c: Creating topic. name: channel:robot_213036, detail: [Aug 18 10:34:28] DEBUG[15089] stasis.c: Topic 'channel:robot_213036': 0x7f0cb01133f0 created [Aug 18 10:34:28] DEBUG[15089] stasis.c: Creating topic. name: cache:669/channel:robot_213036, detail: [Aug 18 10:34:28] DEBUG[15089] stasis.c: Topic 'cache:669/channel:robot_213036': 0x7f0cb010ef90 created [Aug 18 10:34:28] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:28] DEBUG[15090] http.c: match request [ari/channels/213133] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:28] DEBUG[14610] stasis.c: Destroying topic. name: cache:426/channel:213133, detail: [Aug 18 10:34:28] DEBUG[14610] stasis.c: Topic 'cache:426/channel:213133': 0x7f0cb4007560 destroyed [Aug 18 10:34:28] DEBUG[14610] stasis.c: Destroying topic. name: channel:213133, detail: [Aug 18 10:34:28] DEBUG[14610] stasis.c: Topic 'channel:213133': 0x7f0cb40362d0 destroyed [Aug 18 10:34:28] DEBUG[14716] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[15090] http.c: match request [ari/channels/213133] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[14716] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[15083] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 193 instead [Aug 18 10:34:28] DEBUG[15090] http.c: match request [ari/channels/213133] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15090] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[14522] stasis.c: Destroying topic. name: cache:401/channel:1629282853.348, detail: [Aug 18 10:34:28] DEBUG[14522] stasis.c: Topic 'cache:401/channel:1629282853.348': 0x7f0c4005b170 destroyed [Aug 18 10:34:28] DEBUG[14522] stasis.c: Destroying topic. name: channel:1629282853.348, detail: [Aug 18 10:34:28] DEBUG[14522] stasis.c: Topic 'channel:1629282853.348': 0x7f0c400a6b40 destroyed [Aug 18 10:34:28] DEBUG[14716] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14716] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:28] DEBUG[14716] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14716] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be is already using the new technology. [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Finding handler for channels/213133 [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15086] res_ari.c: No explicit handler found for 61075423-3ee2-4d60-8382-ee99e654a5be. Using wildcard bridgeId. [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[14716] channel.c: Channel 0x7f0c200b0230 'Recorder/ARI-00000039;2' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #42 (5) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #42)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864104 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Finding handler for 213133 [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Finding handler for play [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:28] DEBUG[15086] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:28] DEBUG[15086] stasis.c: Creating topic. name: channel:1629282868.584, detail: [Aug 18 10:34:28] DEBUG[15086] stasis.c: Topic 'channel:1629282868.584': 0x7f0ca403b000 created [Aug 18 10:34:28] DEBUG[15086] stasis.c: Creating topic. name: cache:670/channel:1629282868.584, detail: [Aug 18 10:34:28] DEBUG[15086] stasis.c: Topic 'cache:670/channel:1629282868.584': 0x7f0ca4108170 created [Aug 18 10:34:28] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP audio difference is 744, ms is 113 [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking channels create: Didn't match 213133 [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[15090] res_ari.c: Checking channels externalMedia: Didn't match 213133 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: Allocating new SIP dialog for 00ab99953af934b94b85ad8d518c77a2@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15080] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c096b70' [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) RTP allocated port 12398 [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE creating session 0.0.0.0:12398 (12398) [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE create [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE add system candidates [Aug 18 10:34:28] DEBUG[15080] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15080] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE add candidate: 159.65.48.104:12398, 2130706431 [Aug 18 10:34:28] DEBUG[15080] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15080] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE add candidate: 10.131.0.10:12398, 2130706431 [Aug 18 10:34:28] DEBUG[15080] rtp_engine.c: RTP instance '0x7f0c9c096b70' is setup and ready to go [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) ICE stopped [Aug 18 10:34:28] DEBUG[15080] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15080] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15080] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15080] res_rtp_asterisk.c: (0x7f0c9c096b70) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15080] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15080] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15080] chan_sip.c: SIP call-id changed from '00ab99953af934b94b85ad8d518c77a2@127.0.1.1:5060' to '4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15080] stasis.c: Creating topic. name: channel:213203, detail: [Aug 18 10:34:28] DEBUG[15080] stasis.c: Topic 'channel:213203': 0x7f0c9c0aeb10 created [Aug 18 10:34:28] DEBUG[15080] stasis.c: Creating topic. name: cache:671/channel:213203, detail: [Aug 18 10:34:28] DEBUG[15080] stasis.c: Topic 'cache:671/channel:213203': 0x7f0c9c0ae580 created [Aug 18 10:34:28] DEBUG[15090] res_ari.c: No explicit handler found for 213133. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #137 (5) INVITE - 5 [Aug 18 10:34:28] DEBUG[14874] channel.c: Channel 0x7f0c1003abe0 'SIP/zvonobot-000000d7' allocated [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14874] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #137)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116882@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7a74decf Max-Forwards: 70 From: ;tag=as263c5372 To: Contact: Call-ID: 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1952891774 1952891774 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14482 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14873] res_stasis.c: calls_0: Subscribing to 213181 [Aug 18 10:34:28] DEBUG[14873] stasis/app.c: Channel '213181' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14873] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14875] res_stasis.c: calls_0: Subscribing to 213178 [Aug 18 10:34:28] DEBUG[14875] stasis/app.c: Channel '213178' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14873] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[14875] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14846] res_stasis.c: calls_0: Subscribing to 213174 [Aug 18 10:34:28] DEBUG[14846] stasis/app.c: Channel '213174' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Outgoing Call for 79821116866 [Aug 18 10:34:28] DEBUG[14875] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[14846] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14846] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: Allocating new SIP dialog for 791796bf66a19fbc335ce4926f9e48e4@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Outgoing Call for 79821116862 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[15081] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9800dda0' [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963;received=159.65.48.104 From: ;tag=as39d9ed01 To: ;tag=as2c8bd98d Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2bcf9963;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as39d9ed01 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2c8bd98d [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 (Checking To) --From tag as39d9ed01 --To-tag as2c8bd98d [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 16aae03e1dd8482603b67b5f248808a9@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 441 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (4) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (4) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Outgoing Call for 79821116859 [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) RTP allocated port 11540 [Aug 18 10:34:28] DEBUG[14876] channel.c: Channel 0x7f0c1c050150 'SIP/zvonobot-000000d9' allocated [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14876] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE creating session 0.0.0.0:11540 (11540) [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE create [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[14874] res_stasis.c: calls_0: Subscribing to 213180 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (3) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #79 (6) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #79)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116889@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f5df227 Max-Forwards: 70 From: ;tag=as1ac06673 To: Contact: Call-ID: 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 722698992 722698992 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16966 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (3) INVITE - 5 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE add system candidates [Aug 18 10:34:28] DEBUG[14874] stasis/app.c: Channel '213180' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Outgoing Call for 79821116860 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] VERBOSE[15095] chan_sip.c: Audio is at 13594 [Aug 18 10:34:28] VERBOSE[15095] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:28] DEBUG[14874] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14874] http.c: HTTP closing session. Top level [Aug 18 10:34:28] VERBOSE[15095] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: Allocating new SIP dialog for 7905d9db09482827061f11a038182487@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:28] DEBUG[15068] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c80090d20' [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) RTP allocated port 14858 [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE creating session 0.0.0.0:14858 (14858) [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE create [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] VERBOSE[15095] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] DEBUG[15081] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15081] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] VERBOSE[15092] chan_sip.c: Audio is at 19846 [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE add system candidates [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] VERBOSE[15092] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] VERBOSE[15092] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[15092] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Initializing initreq for method INVITE - callid 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116859@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 3 [ 52]: From: ;tag=as0228d9c2 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 6 [ 60]: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] VERBOSE[15092] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #121 [Aug 18 10:34:28] DEBUG[15092] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] VERBOSE[15093] chan_sip.c: Audio is at 18924 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] VERBOSE[15094] chan_sip.c: Audio is at 12308 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Initializing initreq for method INVITE - callid 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116860@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 3 [ 52]: From: ;tag=as17ad889a [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15068] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:28] DEBUG[15068] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE add candidate: 159.65.48.104:14858, 2130706431 [Aug 18 10:34:28] DEBUG[15068] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[15068] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE add candidate: 10.131.0.10:14858, 2130706431 [Aug 18 10:34:28] DEBUG[15068] rtp_engine.c: RTP instance '0x7f0c80090d20' is setup and ready to go [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) ICE stopped [Aug 18 10:34:28] DEBUG[15068] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15068] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[15068] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15068] res_rtp_asterisk.c: (0x7f0c80090d20) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15068] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15068] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15068] chan_sip.c: SIP call-id changed from '7905d9db09482827061f11a038182487@127.0.1.1:5060' to '7766b4206901774644408a490f81d9d2@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15068] stasis.c: Creating topic. name: channel:213196, detail: [Aug 18 10:34:28] DEBUG[15068] stasis.c: Topic 'channel:213196': 0x7f0c800086c0 created [Aug 18 10:34:28] DEBUG[15068] stasis.c: Creating topic. name: cache:672/channel:213196, detail: [Aug 18 10:34:28] DEBUG[15068] stasis.c: Topic 'cache:672/channel:213196': 0x7f0c800323f0 created [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2b2796597f47b4537f2a70003282d682@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE add candidate: 159.65.48.104:11540, 2130706431 [Aug 18 10:34:28] DEBUG[14835] res_rtp_asterisk.c: (0x7f0c40036750) RTCP got report of 76 bytes from 178.62.121.41:10973 [Aug 18 10:34:28] VERBOSE[14835] res_rtp_asterisk.c: 0x7f0c40052640 -- Strict RTP learning complete - Locking on source address 178.62.121.41:10972 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 6 [ 60]: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 [Aug 18 10:34:28] VERBOSE[15092] dial.c: Called zvonobot/79821116859 [Aug 18 10:34:28] VERBOSE[15093] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] VERBOSE[15093] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[15093] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] VERBOSE[15095] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:34:28] DEBUG[15095] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] VERBOSE[15095] dial.c: Called zvonobot/79821116860 [Aug 18 10:34:28] VERBOSE[15094] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] VERBOSE[15094] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[15094] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Initializing initreq for method INVITE - callid 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116866@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 3 [ 52]: From: ;tag=as6d2a22e0 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 6 [ 60]: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] VERBOSE[15094] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #122 [Aug 18 10:34:28] DEBUG[15094] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] VERBOSE[15094] dial.c: Called zvonobot/79821116866 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[15081] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[15081] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE add candidate: 10.131.0.10:11540, 2130706431 [Aug 18 10:34:28] DEBUG[15081] rtp_engine.c: RTP instance '0x7f0c9800dda0' is setup and ready to go [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) ICE stopped [Aug 18 10:34:28] DEBUG[15081] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:28] DEBUG[15081] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS stop [Aug 18 10:34:28] DEBUG[15081] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:28] DEBUG[15081] res_rtp_asterisk.c: (0x7f0c9800dda0) RTCP setup on RTP instance [Aug 18 10:34:28] VERBOSE[15081] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:28] DEBUG[15081] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:28] DEBUG[15081] chan_sip.c: SIP call-id changed from '791796bf66a19fbc335ce4926f9e48e4@127.0.1.1:5060' to '46be811217bc41126929634752a2647e@159.65.48.104:5060' [Aug 18 10:34:28] DEBUG[15081] stasis.c: Creating topic. name: channel:213198, detail: [Aug 18 10:34:28] DEBUG[15081] stasis.c: Topic 'channel:213198': 0x7f0c98046470 created [Aug 18 10:34:28] DEBUG[15081] stasis.c: Creating topic. name: cache:673/channel:213198, detail: [Aug 18 10:34:28] DEBUG[15081] stasis.c: Topic 'cache:673/channel:213198': 0x7f0c98046650 created [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4045900) ICE RTP transport deallocating [Aug 18 10:34:28] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb4045900' [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (2) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Initializing initreq for method INVITE - callid 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116862@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[14876] res_stasis.c: calls_0: Subscribing to 213182 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 [Aug 18 10:34:28] DEBUG[14876] stasis/app.c: Channel '213182' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 3 [ 52]: From: ;tag=as0bc44772 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Outgoing Call for 79821116858 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[14876] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 6 [ 60]: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] VERBOSE[15099] chan_sip.c: Audio is at 17486 [Aug 18 10:34:28] VERBOSE[15099] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] VERBOSE[15093] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #85 [Aug 18 10:34:28] VERBOSE[15099] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 From: ;tag=as563f7715 To: ;tag=as705131ac Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[15093] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] VERBOSE[15099] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Initializing initreq for method INVITE - callid 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116858@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 3 [ 52]: From: ;tag=as2e21b584 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 6 [ 60]: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14876] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] VERBOSE[15099] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #80 [Aug 18 10:34:28] DEBUG[15099] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as563f7715 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as705131ac [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:28] VERBOSE[15099] dial.c: Called zvonobot/79821116858 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[14623] channel.c: Channel 0x7f0c70121770 'UnicastRTP/127.0.0.1:50044-0x7f0c7005f720' allocated [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[14733] res_stasis_playback.c: 1629282854.363: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:28] DEBUG[14733] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:28] DEBUG[14614] channel.c: Channel 0x7f0c3014d000 'Recorder/ARI-00000041;2' allocated [Aug 18 10:34:28] DEBUG[14614] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[14733] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[14761] channel.c: Channel 0x7f0ca4127ea0 'Recorder/ARI-00000047;2' allocated [Aug 18 10:34:28] DEBUG[14761] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:28] VERBOSE[15093] dial.c: Called zvonobot/79821116862 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 (Checking To) --From tag as563f7715 --To-tag as705131ac [Aug 18 10:34:28] DEBUG[15102] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: 0x7f0c3016c860(Recorder/ARI-00000041;2) is joining [Aug 18 10:34:28] DEBUG[15103] bridge_channel.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: 0x7f0ca411d640(Recorder/ARI-00000047;2) is joining [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Stopping retransmission on '30cb159f67289df002568fe9006f4752@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6456ms with no response [Aug 18 10:34:28] DEBUG[14623] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:28] WARNING[20585] chan_sip.c: Hanging up call 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #48 (5) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #48)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235664 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[14779] channel.c: Channel 0x2c9e500 'SIP/zvonobot-000000bb' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000bb - start 1629282861.907605 answer 0.000000 end 1629282868.796006 dur 6.888 bill 1629282868.796 dispo NO ANSWER [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:28] VERBOSE[14623] res_rtp_asterisk.c: 0x7f0c700b7c80 -- Strict RTP learning after remote address set to: 127.0.0.1:50044 [Aug 18 10:34:28] DEBUG[14623] res_stasis.c: calls_0: Subscribing to robot_212967 [Aug 18 10:34:28] DEBUG[15102] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: pushing 0x7f0c3016c860(Recorder/ARI-00000041;2) [Aug 18 10:34:28] DEBUG[14623] stasis/app.c: Channel 'robot_212967' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[15102] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:28] VERBOSE[15102] bridge_channel.c: Channel Recorder/ARI-00000041;2 joined 'simple_bridge' stasis-bridge <5c24e2ba-8671-4745-b349-4500db0d3cb5> [Aug 18 10:34:28] DEBUG[14623] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] VERBOSE[15104] dial.c: Called 127.0.0.1:50044 [Aug 18 10:34:28] VERBOSE[15104] dial.c: UnicastRTP/127.0.0.1:50044-0x7f0c7005f720 answered [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2821bd95 Max-Forwards: 70 From: ;tag=as01310a33 To: ;tag=as5d2e4a10 Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="24cd596d", response="b2c984a34cb4ea164a720eae69997152" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2821bd95 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as01310a33 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as5d2e4a10 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="24cd596d", response="b2c984a34cb4ea164a720eae69997152" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 (Checking From) --From tag as01310a33 --To-tag as5d2e4a10 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:28] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:28] VERBOSE[15104] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50044-0x7f0c7005f720 [Aug 18 10:34:28] DEBUG[15103] bridge_channel.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: pushing 0x7f0ca411d640(Recorder/ARI-00000047;2) [Aug 18 10:34:28] DEBUG[15102] bridge_native_rtp.c: Bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[15102] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[15102] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14879] channel.c: Channel 0x7f0c18018c70 'SIP/zvonobot-000000da' allocated [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:28] DEBUG[15102] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[15102] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[15102] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5 is already using the new technology. [Aug 18 10:34:28] DEBUG[15102] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: 0x7f0c3016c860(Recorder/ARI-00000041;2) is joining simple_bridge technology [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c0801b610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[15103] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:28] VERBOSE[15103] bridge_channel.c: Channel Recorder/ARI-00000047;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:28] DEBUG[14614] res_stasis_recording.c: 1629282858.423: Sending record(213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd.wav) command [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:28] DEBUG[14614] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK2821bd95;received=178.62.121.41 From: ;tag=as01310a33 To: ;tag=as5d2e4a10 Call-ID: 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Setting 0x7f0c1c0b2960(SIP/zvonobot-00000027) state from:0 to:1 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[15104] stasis/app.c: Channel 'robot_212967' is 2 interested in calls_0 [Aug 18 10:34:28] WARNING[14884] app.c: No audio available on Recorder/ARI-0000003e;1?? [Aug 18 10:34:28] VERBOSE[14884] app.c: User hung up [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Session timer stopped: 14 - 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14879] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] DEBUG[14884] res_stasis_recording.c: 1629282858.407: Recording complete [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: pulling 0x7f0c1c0b2960(SIP/zvonobot-00000027) [Aug 18 10:34:28] VERBOSE[14560] bridge_channel.c: Channel SIP/zvonobot-00000027 left 'simple_bridge' stasis-bridge [Aug 18 10:34:28] DEBUG[14884] channel.c: Channel 0x7f0c7c0d8220 'Recorder/ARI-0000003e;1' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c1c0b2960(SIP/zvonobot-00000027) is leaving simple_bridge technology [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772;received=159.65.48.104 From: ;tag=as6c7cfd27 To: ;tag=as7ae5f4b3 Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ca723dd" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6c7cfd27 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ae5f4b3 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ca723dd" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 (Checking To) --From tag as6c7cfd27 --To-tag as7ae5f4b3 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Setting 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) state from:0 to:2 [Aug 18 10:34:28] DEBUG[15105] app.c: play_and_record: , /var/spool/asterisk/recording/213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd, 'wav' [Aug 18 10:34:28] DEBUG[15103] bridge_native_rtp.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d'. Checking compatability for channels 'SIP/zvonobot-00000032' and 'Recorder/ARI-00000047;2' [Aug 18 10:34:28] DEBUG[15103] bridge_native_rtp.c: Bridge 'b9e5c782-ffd8-4b13-9c95-cdea191c152d' can not use native RTP bridge as could not get details [Aug 18 10:34:28] DEBUG[15103] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[15103] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[15103] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[15103] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[15103] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d is already using the new technology. [Aug 18 10:34:28] DEBUG[15103] bridge.c: Bridge b9e5c782-ffd8-4b13-9c95-cdea191c152d: 0x7f0ca411d640(Recorder/ARI-00000047;2) is joining simple_bridge technology [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel Recorder/ARI-00000047;2 setting read format path: slin -> slin [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel SIP/zvonobot-00000032 setting write format path: slin -> ulaw [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel SIP/zvonobot-00000032 setting read format path: ulaw -> slin [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel Recorder/ARI-00000047;2 setting write format path: slin -> slin [Aug 18 10:34:28] DEBUG[15105] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:28] VERBOSE[15105] app.c: x=0, open writing: /var/spool/asterisk/recording/213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd format: wav, 0x7f0c2c0cbb80 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9;received=159.65.48.104 From: ;tag=as5acf84f3 To: ;tag=as08a7808a Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="353dddd4" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5acf84f3 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as08a7808a [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="353dddd4" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 (Checking To) --From tag as5acf84f3 --To-tag as08a7808a [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Stopping retransmission on '5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116897@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK422dd9e9 Max-Forwards: 70 From: ;tag=as5acf84f3 To: Contact: Call-ID: 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #68 (5) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #68)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180528 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14862] channel.c: Channel 0x7f0cac078230 'SIP/zvonobot-000000db' allocated [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:28] DEBUG[14862] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 From: ;tag=as63ca65c0 To: ;tag=as3059a86a Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 446377761 446377761 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3059a86a [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 446377761 446377761 IN IP4 178.62.121.41 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10090 RTP/AVP 0 8 101 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag as3059a86a [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 825 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 From: ;tag=as2d7c4d21 To: ;tag=as6fcc16b3 Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d7c4d21 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6fcc16b3 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 (Checking To) --From tag as2d7c4d21 --To-tag as6fcc16b3 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000027 - start 1629282828.233527 answer 1629282857.606971 end 1629282868.865050 dur 40.631 bill 11.258 dispo ANSWERED [Aug 18 10:34:28] DEBUG[14560] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[14560] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[14560] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14560] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14560] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14560] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06 is already using the new technology. [Aug 18 10:34:28] DEBUG[14761] res_stasis_recording.c: 1629282861.469: Sending record(213014_FacccbxlHWyHfCuOUZlihAHrKMCUFPsZ.wav) command [Aug 18 10:34:28] DEBUG[14761] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:28] DEBUG[14761] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15107] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15107] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:28] DEBUG[15107] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15107] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[15107] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[14878] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: pulling 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) [Aug 18 10:34:28] DEBUG[15107] http.c: Match made with [ari] [Aug 18 10:34:28] VERBOSE[14878] bridge_channel.c: Channel Recorder/ARI-0000003e;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:28] DEBUG[14878] bridge_channel.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06: 0x7f0c7c08ace0(Recorder/ARI-0000003e;2) is leaving simple_bridge technology [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Finding handler for bridges [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Finding handler for bridges [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 From: ;tag=as3a3fa466 To: ;tag=as28515388 Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29914345;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a3fa466 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as28515388 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2069b312" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 (Checking To) --From tag as3a3fa466 --To-tag as28515388 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 67dea07c73c1f69e357cb6190c20cf7e@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (1) INVITE - 5 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 From: ;tag=as563f7715 To: ;tag=as705131ac Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as563f7715 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as705131ac [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 (Checking To) --From tag as563f7715 --To-tag as705131ac [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Stopping retransmission on '30cb159f67289df002568fe9006f4752@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] DEBUG[14878] bridge_native_rtp.c: Bridge 'ef748304-0183-43ff-af76-f67b2d982f06' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[14878] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[14878] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14878] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[14878] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:28] DEBUG[14878] bridge.c: Bridge ef748304-0183-43ff-af76-f67b2d982f06 is already using the new technology. [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Bridge is returning 0x7f0c1c0b2960(SIP/zvonobot-00000027) to read format alaw [Aug 18 10:34:28] DEBUG[15106] app.c: play_and_record: , /var/spool/asterisk/recording/213014_FacccbxlHWyHfCuOUZlihAHrKMCUFPsZ, 'wav' [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel Recorder/ARI-00000047;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:28] DEBUG[15107] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:28] DEBUG[15106] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:28] DEBUG[15103] channel.c: Channel Recorder/ARI-00000047;2 setting write format path: alaw -> slin [Aug 18 10:34:28] DEBUG[15107] stasis.c: Creating topic. name: bridge:4fcf5897-7f85-4abd-99a4-9d663f8db9d4, detail: [Aug 18 10:34:28] DEBUG[15107] stasis.c: Topic 'bridge:4fcf5897-7f85-4abd-99a4-9d663f8db9d4': 0x7f0c340ba6d0 created [Aug 18 10:34:28] DEBUG[15107] stasis.c: Creating topic. name: cache:674/bridge:4fcf5897-7f85-4abd-99a4-9d663f8db9d4, detail: [Aug 18 10:34:28] DEBUG[14878] channel.c: Channel 0x7f0c7c054de0 'Recorder/ARI-0000003e;2' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[14560] channel.c: Channel SIP/zvonobot-00000027 setting read format path: ulaw -> alaw [Aug 18 10:34:28] DEBUG[14862] res_stasis.c: calls_0: Subscribing to 213175 [Aug 18 10:34:28] DEBUG[14560] bridge_channel.c: Bridge is returning 0x7f0c1c0b2960(SIP/zvonobot-00000027) to write format alaw [Aug 18 10:34:28] DEBUG[15107] stasis.c: Topic 'cache:674/bridge:4fcf5897-7f85-4abd-99a4-9d663f8db9d4': 0x7f0c34017520 created [Aug 18 10:34:28] VERBOSE[15106] app.c: x=0, open writing: /var/spool/asterisk/recording/213014_FacccbxlHWyHfCuOUZlihAHrKMCUFPsZ format: wav, 0x7f0c280111c0 [Aug 18 10:34:28] DEBUG[14560] channel.c: Channel SIP/zvonobot-00000027 setting write format path: alaw -> ulaw [Aug 18 10:34:28] DEBUG[15107] bridge_native_rtp.c: Bridge '4fcf5897-7f85-4abd-99a4-9d663f8db9d4' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[14560] stasis/control.c: 213002, ef748304-0183-43ff-af76-f67b2d982f06: Channel was departed from bridge [Aug 18 10:34:28] DEBUG[15107] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[15107] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[15107] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:28] DEBUG[15107] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[14862] stasis/app.c: Channel '213175' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14560] stasis/app.c: bridge 'ef748304-0183-43ff-af76-f67b2d982f06': is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[13141] stasis/control.c: 213002: Channel departing bridge [Aug 18 10:34:28] DEBUG[13141] bridge.c: Waiting for 0x7f0c1c0b2960(SIP/zvonobot-00000027) bridge thread to die. [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:28] DEBUG[14862] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Outgoing Call for 79821116865 [Aug 18 10:34:28] DEBUG[14862] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[15107] bridge.c: Bridge 4fcf5897-7f85-4abd-99a4-9d663f8db9d4: calling simple_bridge technology constructor [Aug 18 10:34:28] DEBUG[15107] bridge.c: Bridge 4fcf5897-7f85-4abd-99a4-9d663f8db9d4: calling simple_bridge technology start [Aug 18 10:34:28] DEBUG[15109] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15107] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14560] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:28] DEBUG[15107] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[13141] stasis/app.c: channel '213002': is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[15109] http.c: HTTP Request URI is /ari/bridges/6199a092-f834-41fb-9e43-7eb7ef40551d/addChannel?channel=1629282858.425%2Crobot_212967 [Aug 18 10:34:28] DEBUG[13141] channel.c: Channel 0x7f0c08023460 'SIP/zvonobot-00000027' hanging up. Refs: 2 [Aug 18 10:34:28] DEBUG[15109] http.c: match request [ari/bridges/6199a092-f834-41fb-9e43-7eb7ef40551d/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:28] VERBOSE[15108] chan_sip.c: Audio is at 16814 [Aug 18 10:34:28] VERBOSE[15108] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:28] VERBOSE[15108] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:28] VERBOSE[15108] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Initializing initreq for method INVITE - callid 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116865@178.62.121.41 SIP/2.0 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 3 [ 52]: From: ;tag=as4bc9e76f [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 6 [ 60]: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:28 GMT [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:28] VERBOSE[15108] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #123 [Aug 18 10:34:28] DEBUG[15108] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK05ab7e00 Max-Forwards: 70 From: ;tag=as26319206 To: ;tag=as08bf07d1 Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="2ef41625", response="be6ef236fe5454101b471c1c93b7254c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:28] DEBUG[15109] http.c: match request [ari/bridges/6199a092-f834-41fb-9e43-7eb7ef40551d/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:28] DEBUG[15109] http.c: match request [ari/bridges/6199a092-f834-41fb-9e43-7eb7ef40551d/addChannel] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15109] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK05ab7e00 [Aug 18 10:34:28] VERBOSE[15108] dial.c: Called zvonobot/79821116865 [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Finding handler for bridges/6199a092-f834-41fb-9e43-7eb7ef40551d/addChannel [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Finding handler for bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:28] DEBUG[15110] http.c: HTTP opening session. Top level [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Finding handler for 6199a092-f834-41fb-9e43-7eb7ef40551d [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as26319206 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as08bf07d1 [Aug 18 10:34:28] DEBUG[15109] res_ari.c: No explicit handler found for 6199a092-f834-41fb-9e43-7eb7ef40551d. Using wildcard bridgeId. [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Finding handler for addChannel [Aug 18 10:34:28] DEBUG[15109] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:28] DEBUG[15109] stasis/control.c: 1629282858.425: Sending channel add_to_bridge command [Aug 18 10:34:28] DEBUG[15110] http.c: HTTP Request URI is /ari/channels/213014/snoop?app=calls_0&spy=in [Aug 18 10:34:28] DEBUG[15110] http.c: match request [ari/channels/213014/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:28] DEBUG[15110] http.c: match request [ari/channels/213014/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="2ef41625", response="be6ef236fe5454101b471c1c93b7254c" [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:28] DEBUG[14617] bridge_roles.c: Roles did not exist on channel Snoop/212967-00000015 [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:28] DEBUG[14617] stasis/control.c: 1629282858.425: Adding to bridge 6199a092-f834-41fb-9e43-7eb7ef40551d [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: = Looking for Call ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 (Checking From) --From tag as26319206 --To-tag as08bf07d1 [Aug 18 10:34:28] DEBUG[15110] http.c: match request [ari/channels/213014/snoop] with handler [ari] len 3 [Aug 18 10:34:28] DEBUG[15110] http.c: Match made with [ari] [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Finding handler for channels/213014/snoop [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14617] stasis/app.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Finding handler for channels [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:28] DEBUG[15111] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: 0x7f0c400a6870(Snoop/212967-00000015) is joining [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:28] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Finding handler for 213014 [Aug 18 10:34:28] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channels create: Didn't match 213014 [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:28] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channels externalMedia: Didn't match 213014 [Aug 18 10:34:28] DEBUG[15110] res_ari.c: No explicit handler found for 213014. Using wildcard channelId. [Aug 18 10:34:28] DEBUG[14893] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:28] DEBUG[14631] chan_sip.c: Hangup call SIP/zvonobot-000000ab, SIP callid 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14631] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:28] DEBUG[14632] chan_sip.c: Hangup call SIP/zvonobot-000000aa, SIP callid 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14632] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:28] DEBUG[14632] res_rtp_asterisk.c: (0x7f0cac0a2c80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[14632] res_rtp_asterisk.c: (0x7f0cac0a2c80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[14632] channel.c: Channel 0x7f0cac090e50 'SIP/zvonobot-000000aa' destroying [Aug 18 10:34:28] DEBUG[14631] res_rtp_asterisk.c: (0x2c83a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[12888] chan_sip.c: Hangup call SIP/zvonobot-00000004, SIP callid 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:28] DEBUG[14893] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[14631] res_rtp_asterisk.c: (0x2c83a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[12888] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[12888] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:28] DEBUG[14631] channel.c: Channel 0x2c950d0 'SIP/zvonobot-000000ab' destroying [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Finding handler for snoop [Aug 18 10:34:28] DEBUG[15111] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: pushing 0x7f0c400a6870(Snoop/212967-00000015) [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282868.588, detail: [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:28] VERBOSE[15111] bridge_channel.c: Channel Snoop/212967-00000015 joined 'simple_bridge' stasis-bridge <6199a092-f834-41fb-9e43-7eb7ef40551d> [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:28] DEBUG[15110] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'channel:1629282868.588': 0x7f0c300b3540 created [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: cache:675/channel:1629282868.588, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:675/channel:1629282868.588': 0x7f0c3008f7b0 created [Aug 18 10:34:28] DEBUG[14879] res_stasis.c: calls_0: Subscribing to 213183 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:28] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:28] DEBUG[15111] bridge_native_rtp.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' can not use native RTP bridge as two channels are required [Aug 18 10:34:28] DEBUG[15111] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:28] DEBUG[15111] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: cache:675/channel:1629282868.588, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:675/channel:1629282868.588': 0x7f0c3008f7b0 destroyed [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282868.588, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'channel:1629282868.588': 0x7f0c300b3540 destroyed [Aug 18 10:34:28] DEBUG[15111] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:28] DEBUG[15111] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:28] DEBUG[15111] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d is already using the new technology. [Aug 18 10:34:28] DEBUG[15111] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: 0x7f0c400a6870(Snoop/212967-00000015) is joining simple_bridge technology [Aug 18 10:34:28] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000aa', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213134', '')] [Aug 18 10:34:28] DEBUG[15109] stasis/control.c: robot_212967: Sending channel add_to_bridge command [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282868.589, detail: [Aug 18 10:34:28] DEBUG[14879] stasis/app.c: Channel '213183' is 1 interested in calls_0 [Aug 18 10:34:28] DEBUG[14617] stasis/app.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' is 2 interested in calls_0 [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'channel:1629282868.589': 0x7f0c300b3540 created [Aug 18 10:34:28] DEBUG[20545] stasis.c: Creating topic. name: cache:676/channel:1629282868.589, detail: [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:676/channel:1629282868.589': 0x7f0c3008f7b0 created [Aug 18 10:34:28] DEBUG[14879] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:28] DEBUG[14879] http.c: HTTP closing session. Top level [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: cache:676/channel:1629282868.589, detail: [Aug 18 10:34:28] DEBUG[15111] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:28] DEBUG[20545] stasis.c: Topic 'cache:676/channel:1629282868.589': 0x7f0c3008f7b0 destroyed [Aug 18 10:34:28] DEBUG[15111] bridge_channel.c: Setting 0x7f0c400a6870(Snoop/212967-00000015) state from:0 to:1 [Aug 18 10:34:28] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282868.589, detail: [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Outgoing Call for 79821116857 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282868.589': 0x7f0c300b3540 destroyed [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000ab', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213137', '')] [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9801aec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[15111] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: pulling 0x7f0c400a6870(Snoop/212967-00000015) [Aug 18 10:34:29] VERBOSE[15111] bridge_channel.c: Channel Snoop/212967-00000015 left 'simple_bridge' stasis-bridge <6199a092-f834-41fb-9e43-7eb7ef40551d> [Aug 18 10:34:29] DEBUG[15111] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: 0x7f0c400a6870(Snoop/212967-00000015) is leaving simple_bridge technology [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '710394295318048c14806fba23b501f2@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK05ab7e00;received=178.62.121.41 From: ;tag=as26319206 To: ;tag=as08bf07d1 Call-ID: 710394295318048c14806fba23b501f2@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (5) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Setting 0x7f0ca0073e00(SIP/zvonobot-00000030) state from:0 to:1 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (5) INVITE - 5 [Aug 18 10:34:29] DEBUG[14904] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212967-00000015 - start 1629282858.711349 answer 1629282858.711349 end 1629282869.012500 dur 10.301 bill 10.301 dispo ANSWERED [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:29] DEBUG[14150] channel.c: Channel 0x7f0ca405f210 'Announcer/ARI-0000002a;2' destroying [Aug 18 10:34:29] DEBUG[14904] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218160 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15111] bridge_native_rtp.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[13624] channel.c: Channel 0x7f0c7c077520 'Recorder/ARI-00000019;2' destroying [Aug 18 10:34:29] VERBOSE[13632] app.c: User hung up [Aug 18 10:34:29] DEBUG[15111] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15111] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13632] res_stasis_recording.c: 1629282837.153: Recording complete [Aug 18 10:34:29] DEBUG[15111] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15111] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session timer stopped: 27 - 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[13632] channel.c: Channel 0x7f0c7c07a200 'Recorder/ARI-00000019;1' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15111] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d is already using the new technology. [Aug 18 10:34:29] VERBOSE[15112] chan_sip.c: Audio is at 10852 [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pulling 0x7f0ca0073e00(SIP/zvonobot-00000030) [Aug 18 10:34:29] VERBOSE[13619] bridge_channel.c: Channel SIP/zvonobot-00000030 left 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0ca0073e00(SIP/zvonobot-00000030) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Setting 0x7f0c24057ae0(Announcer/ARI-00000038;2) state from:0 to:2 [Aug 18 10:34:29] DEBUG[15111] stasis/control.c: 1629282858.425, 6199a092-f834-41fb-9e43-7eb7ef40551d: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[15111] stasis/app.c: bridge '6199a092-f834-41fb-9e43-7eb7ef40551d': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[14617] stasis/control.c: 1629282858.425: Channel departing bridge [Aug 18 10:34:29] VERBOSE[15112] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[13624] stasis.c: Destroying topic. name: cache:185/channel:1629282837.156, detail: [Aug 18 10:34:29] DEBUG[13624] stasis.c: Topic 'cache:185/channel:1629282837.156': 0x7f0c7c01b510 destroyed [Aug 18 10:34:29] DEBUG[13624] stasis.c: Destroying topic. name: channel:1629282837.156, detail: [Aug 18 10:34:29] DEBUG[13624] stasis.c: Topic 'channel:1629282837.156': 0x7f0c7c07c9c0 destroyed [Aug 18 10:34:29] DEBUG[14150] stasis.c: Destroying topic. name: cache:318/channel:1629282847.272, detail: [Aug 18 10:34:29] DEBUG[14150] stasis.c: Topic 'cache:318/channel:1629282847.272': 0x7f0ca40597d0 destroyed [Aug 18 10:34:29] DEBUG[14617] bridge.c: Waiting for 0x7f0c400a6870(Snoop/212967-00000015) bridge thread to die. [Aug 18 10:34:29] DEBUG[14150] stasis.c: Destroying topic. name: channel:1629282847.272, detail: [Aug 18 10:34:29] DEBUG[14150] stasis.c: Topic 'channel:1629282847.272': 0x7f0ca4059600 destroyed [Aug 18 10:34:29] DEBUG[15111] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] VERBOSE[15112] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] VERBOSE[15112] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[14617] stasis/app.c: channel '1629282858.425': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Initializing initreq for method INVITE - callid 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116857@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 3 [ 52]: From: ;tag=as329379c0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 6 [ 60]: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15112] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[14617] stasis/app.c: channel '1629282858.425' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #126 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 From: ;tag=as2ed109a6 To: ;tag=as7faa24c6 Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ef5df89;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ed109a6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7faa24c6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4255d17a" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 (Checking To) --From tag as2ed109a6 --To-tag as7faa24c6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 01fa931654c30892638dda461f55f7f2@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0a2c80) DTLS stop [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0a2c80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0a2c80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac0a2c80) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cac0a2c80' [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15104] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50044-0x7f0c7005f720 [Aug 18 10:34:29] DEBUG[15104] stasis/control.c: robot_212967: Adding to bridge 6199a092-f834-41fb-9e43-7eb7ef40551d [Aug 18 10:34:29] DEBUG[15104] stasis/app.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' is 2 interested in calls_0 [Aug 18 10:34:29] DEBUG[15112] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000030 - start 1629282830.215616 answer 1629282836.641618 end 1629282869.039619 dur 38.824 bill 32.398 dispo ANSWERED [Aug 18 10:34:29] DEBUG[13619] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[13619] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[13619] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13619] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13619] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13619] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 is already using the new technology. [Aug 18 10:34:29] DEBUG[14715] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: pulling 0x7f0c24057ae0(Announcer/ARI-00000038;2) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c83a90) DTLS stop [Aug 18 10:34:29] VERBOSE[14715] bridge_channel.c: Channel Announcer/ARI-00000038;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c83a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14715] bridge_channel.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52: 0x7f0c24057ae0(Announcer/ARI-00000038;2) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[14617] channel.c: Channel 0x7f0c880cb930 'Snoop/212967-00000015' hanging up. Refs: 4 [Aug 18 10:34:29] DEBUG[15113] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: 0x7f0c20008c00(UnicastRTP/127.0.0.1:50044-0x7f0c7005f720) is joining [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c83a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c83a90) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[15113] bridge_channel.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: pushing 0x7f0c20008c00(UnicastRTP/127.0.0.1:50044-0x7f0c7005f720) [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x2c83a90' [Aug 18 10:34:29] VERBOSE[15113] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50044-0x7f0c7005f720 joined 'simple_bridge' stasis-bridge <6199a092-f834-41fb-9e43-7eb7ef40551d> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[14715] bridge_native_rtp.c: Bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:29] DEBUG[14715] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[14715] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14715] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14715] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[14715] bridge.c: Bridge e594e1d1-53fe-4904-8517-472d8e3b8b52 is already using the new technology. [Aug 18 10:34:29] VERBOSE[15112] dial.c: Called zvonobot/79821116857 [Aug 18 10:34:29] DEBUG[15113] bridge_native_rtp.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[14715] channel.c: Channel 0x7f0c2412f860 'Announcer/ARI-00000038;2' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15113] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15113] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15113] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15113] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[15113] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d is already using the new technology. [Aug 18 10:34:29] DEBUG[15113] bridge.c: Bridge 6199a092-f834-41fb-9e43-7eb7ef40551d: 0x7f0c20008c00(UnicastRTP/127.0.0.1:50044-0x7f0c7005f720) is joining simple_bridge technology [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Bridge is returning 0x7f0ca0073e00(SIP/zvonobot-00000030) to read format alaw [Aug 18 10:34:29] DEBUG[13619] channel.c: Channel SIP/zvonobot-00000030 setting read format path: ulaw -> alaw [Aug 18 10:34:29] DEBUG[13619] bridge_channel.c: Bridge is returning 0x7f0ca0073e00(SIP/zvonobot-00000030) to write format alaw [Aug 18 10:34:29] DEBUG[15109] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[15109] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 From: ;tag=as63ca65c0 To: ;tag=as3059a86a Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 446377761 446377761 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 10090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2e817ee5;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as63ca65c0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3059a86a [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 446377761 446377761 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 10090 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 (Checking To) --From tag as63ca65c0 --To-tag as3059a86a [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63c7430f0c6f8039194559375e330124@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 825 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200;received=159.65.48.104 From: ;tag=as154d6f9d To: ;tag=as6670d1e3 Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="588bd726" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as154d6f9d [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6670d1e3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="588bd726" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 (Checking To) --From tag as154d6f9d --To-tag as6670d1e3 [Aug 18 10:34:29] DEBUG[15115] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15115] http.c: HTTP Request URI is /ari/channels/213134 [Aug 18 10:34:29] DEBUG[15115] http.c: match request [ari/channels/213134] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15115] http.c: match request [ari/channels/213134] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116900@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK29af6200 Max-Forwards: 70 From: ;tag=as154d6f9d To: Contact: Call-ID: 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15115] http.c: match request [ari/channels/213134] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:29] DEBUG[15115] http.c: Match made with [ari] [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Finding handler for channels/213134 [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[13619] channel.c: Channel SIP/zvonobot-00000030 setting write format path: alaw -> ulaw [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213134': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[14908] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14633] chan_sip.c: Hangup call SIP/zvonobot-000000ac, SIP callid 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14633] res_rtp_asterisk.c: (0x7f0c24031c50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14633] res_rtp_asterisk.c: (0x7f0c24031c50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14633] channel.c: Channel 0x7f0c24139530 'SIP/zvonobot-000000ac' destroying [Aug 18 10:34:29] DEBUG[12956] chan_sip.c: Hangup call SIP/zvonobot-0000000e, SIP callid 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[12956] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[12956] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.590, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.590': 0x7f0c300fc720 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:677/channel:1629282869.590, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:677/channel:1629282869.590': 0x7f0c30135c60 created [Aug 18 10:34:29] DEBUG[14908] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[13619] stasis/control.c: 213011, e594e1d1-53fe-4904-8517-472d8e3b8b52: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13861] channel.c: Channel 0x7f0c9407a670 'Recorder/ARI-00000024;1' destroying [Aug 18 10:34:29] DEBUG[13852] bridge_channel.c: Setting 0x7f0c940389d0(Recorder/ARI-00000024;2) state from:0 to:1 [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[13861] stasis.c: Destroying topic. name: cache:236/channel:1629282840.195, detail: [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213134' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[13861] stasis.c: Topic 'cache:236/channel:1629282840.195': 0x7f0c94030e90 destroyed [Aug 18 10:34:29] DEBUG[13852] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pulling 0x7f0c940389d0(Recorder/ARI-00000024;2) [Aug 18 10:34:29] VERBOSE[13852] bridge_channel.c: Channel Recorder/ARI-00000024;2 left 'softmix' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:29] DEBUG[13852] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c940389d0(Recorder/ARI-00000024;2) is leaving softmix technology [Aug 18 10:34:29] DEBUG[13852] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677'. Checking compatability for channels 'Announcer/ARI-00000049;2' and 'SIP/zvonobot-00000013' [Aug 18 10:34:29] DEBUG[13852] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as channel 'SIP/zvonobot-00000013' has features which prevent it [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13852] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] VERBOSE[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: switching from softmix technology to simple_bridge [Aug 18 10:34:29] DEBUG[13796] bridge_channel.c: Setting 0x7f0c100f0220(Snoop/212977-0000000b) state from:0 to:1 [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology constructor [Aug 18 10:34:29] DEBUG[13861] stasis.c: Destroying topic. name: channel:1629282840.195, detail: [Aug 18 10:34:29] DEBUG[13861] stasis.c: Topic 'channel:1629282840.195': 0x7f0c94064810 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:677/channel:1629282869.590, detail: [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) to dummy bridge temporarily [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: moving 0x7f0c18091350(SIP/zvonobot-00000013) to dummy bridge temporarily [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:677/channel:1629282869.590': 0x7f0c30135c60 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.590, detail: [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) is leaving softmix technology (dummy) [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:436/channel:213134, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:436/channel:213134': 0x7f0cac03ffe0 destroyed [Aug 18 10:34:29] DEBUG[15104] stasis/app.c: Bridge '6199a092-f834-41fb-9e43-7eb7ef40551d' is 3 interested in calls_0 [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is leaving softmix technology (dummy) [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.590': 0x7f0c300fc720 destroyed [Aug 18 10:34:29] DEBUG[13619] stasis/app.c: bridge 'e594e1d1-53fe-4904-8517-472d8e3b8b52': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000ac', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213141', '')] [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology stop [Aug 18 10:34:29] DEBUG[13201] stasis/control.c: 213011: Channel departing bridge [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) is joining simple_bridge technology [Aug 18 10:34:29] DEBUG[13201] bridge.c: Waiting for 0x7f0ca0073e00(SIP/zvonobot-00000030) bridge thread to die. [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[13852] channel.c: Channel Announcer/ARI-00000049;2 setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[13852] channel.c: Channel Announcer/ARI-00000049;2 setting read format path: slin -> slin [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is joining simple_bridge technology [Aug 18 10:34:29] DEBUG[13852] channel.c: Channel Announcer/ARI-00000049;2 setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[13852] channel.c: Channel Announcer/ARI-00000049;2 setting read format path: slin -> slin [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling simple_bridge technology start [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[13619] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: deferring softmix technology destructor [Aug 18 10:34:29] DEBUG[13852] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: queueing action type:13 sub:1000 [Aug 18 10:34:29] DEBUG[13201] stasis/app.c: channel '213011': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[13201] channel.c: Channel 0x7f0c980222e0 'SIP/zvonobot-00000030' hanging up. Refs: 3 [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Finding handler for 213134 [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking channels create: Didn't match 213134 [Aug 18 10:34:29] DEBUG[13796] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: pulling 0x7f0c100f0220(Snoop/212977-0000000b) [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213134, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213134': 0x7f0cac047f60 destroyed [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:29] VERBOSE[13796] bridge_channel.c: Channel Snoop/212977-0000000b left 'simple_bridge' stasis-bridge <48086187-3f40-424c-b978-0d6c6da7141b> [Aug 18 10:34:29] DEBUG[13796] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c100f0220(Snoop/212977-0000000b) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[14644] chan_sip.c: Hangup call SIP/zvonobot-000000ae, SIP callid 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14644] res_rtp_asterisk.c: (0x7f0c180962b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14644] res_rtp_asterisk.c: (0x7f0c180962b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14162] channel.c: Channel 0x7f0ca4085510 'Announcer/ARI-0000002a;1' destroying [Aug 18 10:34:29] DEBUG[14646] chan_sip.c: Hangup call SIP/zvonobot-000000ad, SIP callid 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14646] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 From: ;tag=as76ba9cfd To: ;tag=as041dcfd6 Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[14646] res_rtp_asterisk.c: (0x7f0cb4070230) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14646] res_rtp_asterisk.c: (0x7f0cb4070230) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14646] channel.c: Channel 0x7f0cb407fe90 'SIP/zvonobot-000000ad' destroying [Aug 18 10:34:29] DEBUG[14644] channel.c: Channel 0x7f0c1807ed80 'SIP/zvonobot-000000ae' destroying [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as76ba9cfd [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212977-0000000b - start 1629282839.866018 answer 1629282839.866018 end 1629282869.114705 dur 29.248 bill 29.248 dispo ANSWERED [Aug 18 10:34:29] DEBUG[15116] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[14785] channel.c: Channel 0x7f0c28096950 'SIP/zvonobot-000000ce' allocated [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14785] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as041dcfd6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 (Checking To) --From tag as76ba9cfd --To-tag as041dcfd6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 29b2374b344692165b25e9de23da23d4@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '29b2374b344692165b25e9de23da23d4@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24031c50) DTLS stop [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24031c50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24031c50) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24031c50) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[15116] http.c: HTTP Request URI is /ari/channels/213137 [Aug 18 10:34:29] DEBUG[14162] stasis.c: Destroying topic. name: cache:298/channel:1629282846.255, detail: [Aug 18 10:34:29] DEBUG[14162] stasis.c: Topic 'cache:298/channel:1629282846.255': 0x7f0ca4043b10 destroyed [Aug 18 10:34:29] DEBUG[14162] stasis.c: Destroying topic. name: channel:1629282846.255, detail: [Aug 18 10:34:29] DEBUG[14162] stasis.c: Topic 'channel:1629282846.255': 0x7f0ca406c5c0 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.591, detail: [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c24031c50' [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213137': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[15116] http.c: match request [ari/channels/213137] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.591': 0x7f0c300e2080 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:678/channel:1629282869.591, detail: [Aug 18 10:34:29] DEBUG[15115] res_ari.c: Checking channels externalMedia: Didn't match 213134 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:678/channel:1629282869.591': 0x7f0c300a9240 created [Aug 18 10:34:29] DEBUG[15115] res_ari.c: No explicit handler found for 213134. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213137' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[13796] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[15116] http.c: match request [ari/channels/213137] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:437/channel:213137, detail: [Aug 18 10:34:29] DEBUG[13796] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[14785] res_stasis.c: calls_0: Subscribing to 213170 [Aug 18 10:34:29] DEBUG[15116] http.c: match request [ari/channels/213137] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[13796] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:437/channel:213137': 0x2c97b70 destroyed [Aug 18 10:34:29] DEBUG[15116] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[14929] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[13796] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13796] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13796] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b is already using the new technology. [Aug 18 10:34:29] DEBUG[14929] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face;received=159.65.48.104 From: ;tag=as1a5706e7 To: ;tag=as6aac0e8c Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e47d332" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:678/channel:1629282869.591, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:678/channel:1629282869.591': 0x7f0c300a9240 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.591, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.591': 0x7f0c300e2080 destroyed [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Finding handler for channels/213137 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1a5706e7 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6aac0e8c [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000ad', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213136', '')] [Aug 18 10:34:29] DEBUG[14785] stasis/app.c: Channel '213170' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e47d332" [Aug 18 10:34:29] DEBUG[14939] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: stopping mixing thread [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:29] DEBUG[20534] bridge_softmix.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: Waiting for mixing thread to die. [Aug 18 10:34:29] DEBUG[14946] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:29] DEBUG[13704] channel.c: SIP/zvonobot-00000013: Dropping redundant connected line update "" <>. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.592, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.592': 0x7f0c300e2080 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:679/channel:1629282869.592, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:679/channel:1629282869.592': 0x7f0c300a9240 created [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213137, detail: [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Finding handler for channels [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 (Checking To) --From tag as1a5706e7 --To-tag as6aac0e8c [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213137': 0x2c97140 destroyed [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14785] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:679/channel:1629282869.592, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:679/channel:1629282869.592': 0x7f0c300a9240 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.592, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.592': 0x7f0c300e2080 destroyed [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7366b47a3b83dfbf33109b684b63ddf9@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180962b0) DTLS stop [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000ae', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213135', '')] [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180962b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[13852] channel.c: Channel 0x7f0c9400a450 'Recorder/ARI-00000024;2' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Outgoing Call for 79821116870 [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[13796] bridge_channel.c: Bridge is returning 0x7f0c100f0220(Snoop/212977-0000000b) to read format slin [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180962b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14785] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180962b0) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[13796] channel.c: Channel Snoop/212977-0000000b setting read format path: slin -> slin [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c180962b0' [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[13796] bridge_channel.c: Bridge is returning 0x7f0c100f0220(Snoop/212977-0000000b) to write format slin [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4070230) DTLS stop [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4070230) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[13796] channel.c: Channel Snoop/212977-0000000b setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[13796] stasis/control.c: 1629282839.183, 48086187-3f40-424c-b978-0d6c6da7141b: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13796] stasis/app.c: bridge '48086187-3f40-424c-b978-0d6c6da7141b': is 3 interested in calls_0 [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Finding handler for 213137 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4070230) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4070230) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb4070230' [Aug 18 10:34:29] DEBUG[14934] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20' [Aug 18 10:34:29] DEBUG[14856] channel.c: Channel 0x7f0c8c10bad0 'Announcer/ARI-0000003d;1' destroying [Aug 18 10:34:29] DEBUG[13672] channel.c: Channel 0x7f0c7007bb40 'Announcer/ARI-0000001b;2' destroying [Aug 18 10:34:29] DEBUG[13672] stasis.c: Destroying topic. name: cache:206/channel:1629282838.173, detail: [Aug 18 10:34:29] DEBUG[13672] stasis.c: Topic 'cache:206/channel:1629282838.173': 0x7f0c7005b740 destroyed [Aug 18 10:34:29] DEBUG[13672] stasis.c: Destroying topic. name: channel:1629282838.173, detail: [Aug 18 10:34:29] DEBUG[13672] stasis.c: Topic 'channel:1629282838.173': 0x7f0c7007cb00 destroyed [Aug 18 10:34:29] DEBUG[13686] channel.c: Channel 0x7f0c7c0282f0 'SIP/zvonobot-0000002b' destroying [Aug 18 10:34:29] DEBUG[13686] channel.c: Channel 0x7f0ca0076bd0 'Snoop/213008-0000000a' destroying [Aug 18 10:34:29] DEBUG[13686] stasis.c: Destroying topic. name: cache:215/channel:1629282839.182, detail: [Aug 18 10:34:29] DEBUG[13686] stasis.c: Topic 'cache:215/channel:1629282839.182': 0x7f0ca0022980 destroyed [Aug 18 10:34:29] DEBUG[13686] stasis.c: Destroying topic. name: channel:1629282839.182, detail: [Aug 18 10:34:29] DEBUG[13686] stasis.c: Topic 'channel:1629282839.182': 0x7f0ca00311d0 destroyed [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking channels create: Didn't match 213137 [Aug 18 10:34:29] DEBUG[14659] chan_sip.c: Hangup call SIP/zvonobot-000000b1, SIP callid 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[13114] chan_sip.c: Hangup call SIP/zvonobot-00000020, SIP callid 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[13114] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[13114] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14674] chan_sip.c: Hangup call SIP/zvonobot-000000b2, SIP callid 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14856] stasis.c: Destroying topic. name: cache:467/channel:1629282858.405, detail: [Aug 18 10:34:29] DEBUG[14856] stasis.c: Topic 'cache:467/channel:1629282858.405': 0x7f0c8c118d00 destroyed [Aug 18 10:34:29] DEBUG[14856] stasis.c: Destroying topic. name: channel:1629282858.405, detail: [Aug 18 10:34:29] DEBUG[14856] stasis.c: Topic 'channel:1629282858.405': 0x7f0c8c06ade0 destroyed [Aug 18 10:34:29] DEBUG[14680] chan_sip.c: Hangup call SIP/zvonobot-000000b3, SIP callid 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14659] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[14674] res_rtp_asterisk.c: (0x7f0c30041010) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[13706] stasis/control.c: 1629282839.183: Channel departing bridge [Aug 18 10:34:29] DEBUG[13289] chan_sip.c: Hangup call SIP/zvonobot-0000003a, SIP callid 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14659] res_rtp_asterisk.c: (0x7f0c280e6850) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14659] res_rtp_asterisk.c: (0x7f0c280e6850) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14659] channel.c: Channel 0x7f0c280eeac0 'SIP/zvonobot-000000b1' destroying [Aug 18 10:34:29] DEBUG[13796] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[14680] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:29] DEBUG[14680] res_rtp_asterisk.c: (0x7f0c20064fb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14680] res_rtp_asterisk.c: (0x7f0c20064fb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14680] channel.c: Channel 0x7f0c200b11b0 'SIP/zvonobot-000000b3' destroying [Aug 18 10:34:29] DEBUG[14674] res_rtp_asterisk.c: (0x7f0c30041010) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14844] bridge_channel.c: Setting 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) state from:0 to:1 [Aug 18 10:34:29] DEBUG[14844] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: pulling 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) [Aug 18 10:34:29] VERBOSE[14844] bridge_channel.c: Channel Announcer/ARI-0000003d;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[14844] bridge_channel.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846: 0x7f0c8c01d4d0(Announcer/ARI-0000003d;2) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[14844] bridge_native_rtp.c: Bridge 'fbfe71c6-df7c-4b8c-8b66-37207aaf9846' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[14844] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[14844] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14844] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14844] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[14844] bridge.c: Bridge fbfe71c6-df7c-4b8c-8b66-37207aaf9846 is already using the new technology. [Aug 18 10:34:29] DEBUG[14844] channel.c: Channel 0x7f0c8c034ba0 'Announcer/ARI-0000003d;2' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[13443] channel.c: Channel 0x7f0c18087070 'Recorder/ARI-00000011;2' destroying [Aug 18 10:34:29] DEBUG[13443] stasis.c: Destroying topic. name: cache:147/channel:1629282835.123, detail: [Aug 18 10:34:29] DEBUG[13443] stasis.c: Topic 'cache:147/channel:1629282835.123': 0x7f0c1800e6f0 destroyed [Aug 18 10:34:29] DEBUG[13443] stasis.c: Destroying topic. name: channel:1629282835.123, detail: [Aug 18 10:34:29] DEBUG[13443] stasis.c: Topic 'channel:1629282835.123': 0x7f0c18093e40 destroyed [Aug 18 10:34:29] VERBOSE[13444] app.c: User hung up [Aug 18 10:34:29] DEBUG[13444] res_stasis_recording.c: 1629282835.122: Recording complete [Aug 18 10:34:29] DEBUG[13444] channel.c: Channel 0x7f0c180acf90 'Recorder/ARI-00000011;1' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[13706] bridge.c: Waiting for 0x7f0c100f0220(Snoop/212977-0000000b) bridge thread to die. [Aug 18 10:34:29] DEBUG[13289] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[14674] channel.c: Channel 0x7f0c300fa180 'SIP/zvonobot-000000b2' destroying [Aug 18 10:34:29] DEBUG[13706] stasis/app.c: channel '1629282839.183': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[13706] stasis/app.c: channel '1629282839.183' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[13706] channel.c: Channel 0x7f0ca800d0a0 'Snoop/212977-0000000b' hanging up. Refs: 3 [Aug 18 10:34:29] DEBUG[13466] bridge_channel.c: Setting 0x7f0c34027b30(Snoop/212995-00000007) state from:0 to:1 [Aug 18 10:34:29] DEBUG[13289] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[13289] channel.c: Channel 0x7f0c700595e0 'SIP/zvonobot-0000003a' destroying [Aug 18 10:34:29] DEBUG[15116] res_ari.c: Checking channels externalMedia: Didn't match 213137 [Aug 18 10:34:29] DEBUG[14934] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[14934] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15116] res_ari.c: No explicit handler found for 213137. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[13799] bridge_channel.c: Setting 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) state from:0 to:1 [Aug 18 10:34:29] DEBUG[13799] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: pulling 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) [Aug 18 10:34:29] VERBOSE[13799] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 left 'simple_bridge' stasis-bridge <48086187-3f40-424c-b978-0d6c6da7141b> [Aug 18 10:34:29] DEBUG[13799] bridge_channel.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b: 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[13799] bridge_native_rtp.c: Bridge '48086187-3f40-424c-b978-0d6c6da7141b' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[13799] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[13799] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13799] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13799] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13799] bridge.c: Bridge 48086187-3f40-424c-b978-0d6c6da7141b is already using the new technology. [Aug 18 10:34:29] DEBUG[15118] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15118] http.c: HTTP Request URI is /ari/channels/212977 [Aug 18 10:34:29] DEBUG[13466] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: pulling 0x7f0c34027b30(Snoop/212995-00000007) [Aug 18 10:34:29] VERBOSE[13466] bridge_channel.c: Channel Snoop/212995-00000007 left 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[13466] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c34027b30(Snoop/212995-00000007) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[14641] res_rtp_asterisk.c: (0x7f0c18094150) RTCP got report of 76 bytes from 178.62.121.41:17731 [Aug 18 10:34:29] DEBUG[13466] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[13466] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[13799] bridge_channel.c: Bridge is returning 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) to write format slin16 [Aug 18 10:34:29] DEBUG[13799] channel.c: Channel UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20 setting write format path: slin16 -> slin16 [Aug 18 10:34:29] DEBUG[13799] stasis/control.c: robot_212977, 48086187-3f40-424c-b978-0d6c6da7141b: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13466] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13466] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15118] http.c: match request [ari/channels/212977] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[13799] stasis/app.c: bridge '48086187-3f40-424c-b978-0d6c6da7141b': is 2 interested in calls_0 [Aug 18 10:34:29] DEBUG[15118] http.c: match request [ari/channels/212977] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[13466] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13466] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5 is already using the new technology. [Aug 18 10:34:29] DEBUG[15118] http.c: match request [ari/channels/212977] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15118] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[13777] stasis/control.c: robot_212977: Channel departing bridge [Aug 18 10:34:29] DEBUG[13466] bridge_channel.c: Bridge is returning 0x7f0c34027b30(Snoop/212995-00000007) to read format slin [Aug 18 10:34:29] DEBUG[13777] bridge.c: Waiting for 0x7f0c3c05f8e0(UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20) bridge thread to die. [Aug 18 10:34:29] DEBUG[13466] channel.c: Channel Snoop/212995-00000007 setting read format path: slin -> slin [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.593, detail: [Aug 18 10:34:29] DEBUG[13466] bridge_channel.c: Bridge is returning 0x7f0c34027b30(Snoop/212995-00000007) to write format slin [Aug 18 10:34:29] DEBUG[13466] channel.c: Channel Snoop/212995-00000007 setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Finding handler for channels/212977 [Aug 18 10:34:29] DEBUG[13799] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9;received=159.65.48.104 From: ;tag=as03ee25b2 To: ;tag=as33b73ba9 Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="725cfcb5" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4f5adcb9;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as03ee25b2 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as33b73ba9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="725cfcb5" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 (Checking To) --From tag as03ee25b2 --To-tag as33b73ba9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 07f82ab44968293544eb273a476d91c1@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[13466] stasis/control.c: 1629282835.124, d177377e-a80b-4ad9-826a-cece7d5abce5: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13466] stasis/app.c: bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5': is 3 interested in calls_0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.593': 0x7f0c300414a0 created [Aug 18 10:34:29] DEBUG[13447] stasis/control.c: 1629282835.124: Channel departing bridge [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[13447] bridge.c: Waiting for 0x7f0c34027b30(Snoop/212995-00000007) bridge thread to die. [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[13777] stasis/app.c: channel 'robot_212977': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:680/channel:1629282869.593, detail: [Aug 18 10:34:29] DEBUG[13777] channel.c: Channel 0x7f0c1c120cb0 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[13466] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:680/channel:1629282869.593': 0x7f0c300b9f10 created [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1af633b070d358211c7d70633c29ee6c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[13447] stasis/app.c: channel '1629282835.124': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[13447] stasis/app.c: channel '1629282835.124' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[13447] channel.c: Channel 0x7f0c280c6fb0 'Snoop/212995-00000007' hanging up. Refs: 3 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280e6850) DTLS stop [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280e6850) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] VERBOSE[15117] chan_sip.c: Audio is at 14652 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280e6850) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:680/channel:1629282869.593, detail: [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c280e6850) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:680/channel:1629282869.593': 0x7f0c300b9f10 destroyed [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c280e6850' [Aug 18 10:34:29] VERBOSE[15117] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Finding handler for 212977 [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking channels create: Didn't match 212977 [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15118] res_ari.c: Checking channels externalMedia: Didn't match 212977 [Aug 18 10:34:29] DEBUG[15118] res_ari.c: No explicit handler found for 212977. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '7496c2db547867277094cc9977c08d0c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20064fb0) DTLS stop [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20064fb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.593, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.593': 0x7f0c300414a0 destroyed [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20064fb0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20064fb0) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c20064fb0' [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000002b', '', 'Stasis', 'calls_0', 32, 26, 'ANSWERED', 3, '', '213008', '')] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5bf058b45a6b86266714541f697c9d35@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30041010) DTLS stop [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30041010) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] VERBOSE[15117] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30041010) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] VERBOSE[15117] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.594, detail: [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30041010) ICE RTP transport deallocating [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Initializing initreq for method INVITE - callid 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.594': 0x7f0c300414a0 created [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116870@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:681/channel:1629282869.594, detail: [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c30041010' [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 3 [ 52]: From: ;tag=as36bdbaac [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 6 [ 60]: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15117] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:681/channel:1629282869.594': 0x7f0c30028b90 created [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #115 [Aug 18 10:34:29] DEBUG[15117] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[15117] dial.c: Called zvonobot/79821116870 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:681/channel:1629282869.594, detail: [Aug 18 10:34:29] DEBUG[14936] channel.c: Channel 0x7f0c2c049750 'SIP/zvonobot-000000dc' allocated [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:681/channel:1629282869.594': 0x7f0c30028b90 destroyed [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14936] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[14936] res_stasis.c: calls_0: Subscribing to 213184 [Aug 18 10:34:29] DEBUG[14936] stasis/app.c: Channel '213184' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Outgoing Call for 79821116856 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[14936] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[14936] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[14959] channel.c: Channel 0x7f0c8c123110 'SIP/zvonobot-000000dd' allocated [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6;received=159.65.48.104 From: ;tag=as7b595413 To: ;tag=as0da508ae Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c6e0415" Content-Length: 0 <-------------> [Aug 18 10:34:29] VERBOSE[15120] chan_sip.c: Audio is at 12150 [Aug 18 10:34:29] VERBOSE[15120] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14959] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6;received=159.65.48.104 [Aug 18 10:34:29] VERBOSE[15120] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.594, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.594': 0x7f0c300414a0 destroyed [Aug 18 10:34:29] VERBOSE[15120] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Initializing initreq for method INVITE - callid 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116856@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 3 [ 52]: From: ;tag=as6b79f1a3 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 6 [ 60]: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15120] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:34:29] DEBUG[15120] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[15120] dial.c: Called zvonobot/79821116856 [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <>', '', 's', 'default', 'Snoop/213008-0000000a', 'UnicastRTP/127.0.0.1:50118-0x7f0ca804bf40', 'Stasis', 'calls_0', 16, 16, 'ANSWERED', 3, '', '1629282839.182', '')] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7b595413 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0da508ae [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c6e0415" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 (Checking To) --From tag as7b595413 --To-tag as0da508ae [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116887@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK115bedb6 Max-Forwards: 70 From: ;tag=as7b595413 To: Contact: Call-ID: 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #13 (5) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #13)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754311 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (5) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028039 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14959] res_stasis.c: calls_0: Subscribing to 213192 [Aug 18 10:34:29] DEBUG[14959] stasis/app.c: Channel '213192' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[14959] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Outgoing Call for 79821116848 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.595, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.595': 0x7f0c3003b930 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:682/channel:1629282869.595, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:682/channel:1629282869.595': 0x7f0c300abff0 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:682/channel:1629282869.595, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:682/channel:1629282869.595': 0x7f0c300abff0 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.595, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.595': 0x7f0c3003b930 destroyed [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b1', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213142', '')] [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[14959] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 From: ;tag=as10d8c0eb To: ;tag=as70da9059 Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as10d8c0eb [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70da9059 [Aug 18 10:34:29] DEBUG[15121] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15121] http.c: HTTP Request URI is /ari/channels/213141 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[15122] chan_sip.c: Audio is at 11040 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] VERBOSE[15122] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15121] http.c: match request [ari/channels/213141] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 (Checking To) --From tag as10d8c0eb --To-tag as70da9059 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213141': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] VERBOSE[15122] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.596, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.596': 0x7f0c300571f0 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:683/channel:1629282869.596, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:683/channel:1629282869.596': 0x7f0c300abff0 created [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:683/channel:1629282869.596, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:683/channel:1629282869.596': 0x7f0c300abff0 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.596, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.596': 0x7f0c300571f0 destroyed [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b2', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213143', '')] [Aug 18 10:34:29] DEBUG[15121] http.c: match request [ari/channels/213141] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (6) INVITE - 5 [Aug 18 10:34:29] DEBUG[15121] http.c: match request [ari/channels/213141] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116893@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2b67a976 Max-Forwards: 70 From: ;tag=as2846b9c9 To: Contact: Call-ID: 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 147948068 147948068 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17200 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213141' unsubscribed from calls_0 [Aug 18 10:34:29] VERBOSE[15122] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.597, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.597': 0x7f0c3003b930 created [Aug 18 10:34:29] DEBUG[14956] channel.c: Channel 0x7f0c8409c6d0 'SIP/zvonobot-000000de' allocated [Aug 18 10:34:29] DEBUG[15121] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:684/channel:1629282869.597, detail: [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:684/channel:1629282869.597': 0x7f0c300abff0 created [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14956] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b;received=159.65.48.104 From: ;tag=as3a1d6e7b To: ;tag=as6e4b6af6 Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27bdcc69" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK64773f6b;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a1d6e7b [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Initializing initreq for method INVITE - callid 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6e4b6af6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27bdcc69" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 (Checking To) --From tag as3a1d6e7b --To-tag as6e4b6af6 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7496c2db547867277094cc9977c08d0c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as7e2e6628 Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1817551672 1817551672 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15352 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7e2e6628 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Finding handler for channels/213141 [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:29] DEBUG[14956] res_stasis.c: calls_0: Subscribing to 213190 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:684/channel:1629282869.597, detail: [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116848@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1817551672 1817551672 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:684/channel:1629282869.597': 0x7f0c300abff0 destroyed [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.597, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15352 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as7e2e6628 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.597': 0x7f0c3003b930 destroyed [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Finding handler for 213141 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:438/channel:213141, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (4) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:438/channel:213141': 0x7f0c24140130 destroyed [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 3 [ 52]: From: ;tag=as517a3e83 [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking channels create: Didn't match 213141 [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15121] res_ari.c: Checking channels externalMedia: Didn't match 213141 [Aug 18 10:34:29] DEBUG[15121] res_ari.c: No explicit handler found for 213141. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000003a', '', 'Stasis', 'calls_0', 33, 8, 'ANSWERED', 3, '', '213021', '')] [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 6 [ 60]: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[14956] stasis/app.c: Channel '213190' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15122] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[14956] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #27 [Aug 18 10:34:29] DEBUG[15122] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213141, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213141': 0x7f0c240ed3e0 destroyed [Aug 18 10:34:29] DEBUG[14956] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9;received=159.65.48.104 From: ;tag=as6ac21020 To: ;tag=as01e0c440 Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="346d51ec" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b3657b9;received=159.65.48.104 [Aug 18 10:34:29] VERBOSE[15122] dial.c: Called zvonobot/79821116848 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282869.598, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6ac21020 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as01e0c440 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.598': 0x7f0c300571f0 created [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Outgoing Call for 79821116850 [Aug 18 10:34:29] WARNING[14899] app.c: No audio available on Recorder/ARI-00000040;1?? [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:29] VERBOSE[14899] app.c: User hung up [Aug 18 10:34:29] DEBUG[20545] stasis.c: Creating topic. name: cache:685/channel:1629282869.598, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[14899] res_stasis_recording.c: 1629282858.413: Recording complete [Aug 18 10:34:29] DEBUG[14899] channel.c: Channel 0x7f0c10106460 'Recorder/ARI-00000040;1' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:685/channel:1629282869.598': 0x7f0c300abff0 created [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="346d51ec" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 (Checking To) --From tag as6ac21020 --To-tag as01e0c440 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7d61341d056ac27e2031cc405530ac7b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[15123] chan_sip.c: Audio is at 17618 [Aug 18 10:34:29] VERBOSE[15123] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] VERBOSE[15123] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] VERBOSE[15123] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 From: ;tag=as1c2a52a2 To: ;tag=as025c937f Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15124] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Initializing initreq for method INVITE - callid 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1c2a52a2 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116850@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15124] http.c: HTTP Request URI is /ari/playbacks/869f7bee-0739-41c3-a7ce-b82939e8277b [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as025c937f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 3 [ 52]: From: ;tag=as09a15a28 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15124] http.c: match request [ari/playbacks/869f7bee-0739-41c3-a7ce-b82939e8277b] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 6 [ 60]: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15124] http.c: match request [ari/playbacks/869f7bee-0739-41c3-a7ce-b82939e8277b] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15124] http.c: match request [ari/playbacks/869f7bee-0739-41c3-a7ce-b82939e8277b] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: cache:685/channel:1629282869.598, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'cache:685/channel:1629282869.598': 0x7f0c300abff0 destroyed [Aug 18 10:34:29] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282869.598, detail: [Aug 18 10:34:29] DEBUG[20545] stasis.c: Topic 'channel:1629282869.598': 0x7f0c300571f0 destroyed [Aug 18 10:34:29] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b3', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213139', '')] [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 (Checking To) --From tag as1c2a52a2 --To-tag as025c937f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15124] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (5) BYE - 8 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: BYE sip:79821117032@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12 Max-Forwards: 70 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117032@178.62.121.41:5060", nonce="7c4d0d10", response="973c2e5b2634ba9cd88002faf1c920c6" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[15123] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #113 [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Finding handler for playbacks/869f7bee-0739-41c3-a7ce-b82939e8277b [Aug 18 10:34:29] DEBUG[15123] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487;received=159.65.48.104 From: ;tag=as123f2352 To: ;tag=as0ec69af0 Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a004025" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as123f2352 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0ec69af0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a004025" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 (Checking To) --From tag as123f2352 --To-tag as0ec69af0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a;received=159.65.48.104 From: ;tag=as148e9e8b To: ;tag=as702f5751 Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b6a9896" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as148e9e8b [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as702f5751 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b6a9896" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 (Checking To) --From tag as148e9e8b --To-tag as702f5751 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116890@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK54fe0a8a Max-Forwards: 70 From: ;tag=as148e9e8b To: Contact: Call-ID: 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Finding handler for playbacks [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Finding handler for 869f7bee-0739-41c3-a7ce-b82939e8277b [Aug 18 10:34:29] DEBUG[15124] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15124] res_ari.c: No explicit handler found for 869f7bee-0739-41c3-a7ce-b82939e8277b. Using wildcard playbackId. [Aug 18 10:34:29] DEBUG[15124] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[15124] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772;received=159.65.48.104 From: ;tag=as6c7cfd27 To: ;tag=as7ae5f4b3 Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ca723dd" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK37c50772;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6c7cfd27 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ae5f4b3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212995-00000007 - start 1629282835.319027 answer 1629282835.319027 end 1629282869.214942 dur 33.895 bill 33.895 dispo ANSWERED [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ca723dd" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 (Checking To) --From tag as6c7cfd27 --To-tag as7ae5f4b3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2b2796597f47b4537f2a70003282d682@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[15125] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213136': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213136' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[15126] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:439/channel:213136, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:439/channel:213136': 0x7f0cb40921c0 destroyed [Aug 18 10:34:29] DEBUG[15125] http.c: HTTP Request URI is /ari/channels/robot_213011 [Aug 18 10:34:29] VERBOSE[15123] dial.c: Called zvonobot/79821116850 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348;received=159.65.48.104 From: ;tag=as67ede665 To: ;tag=as495e7ce0 Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a032c56" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as67ede665 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as495e7ce0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a032c56" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 (Checking To) --From tag as67ede665 --To-tag as495e7ce0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116896@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348 Max-Forwards: 70 From: ;tag=as67ede665 To: Contact: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15127] http.c: HTTP opening session. Top level [Aug 18 10:34:29] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213135': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213135' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[15127] http.c: HTTP Request URI is /ari/channels/213135 [Aug 18 10:34:29] DEBUG[15126] http.c: HTTP Request URI is /ari/channels/213136 [Aug 18 10:34:29] WARNING[20585] chan_sip.c: Hanging up call 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:442/channel:213135, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:442/channel:213135': 0x7f0c180c8af0 destroyed [Aug 18 10:34:29] DEBUG[15125] http.c: match request [ari/channels/robot_213011] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213135, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213135': 0x7f0c18096a40 destroyed [Aug 18 10:34:29] DEBUG[14789] channel.c: Channel 0x7f0c38014ce0 'SIP/zvonobot-000000cf' allocated [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14789] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[15127] http.c: match request [ari/channels/213135] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15125] http.c: match request [ari/channels/robot_213011] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15126] http.c: match request [ari/channels/213136] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[14802] channel.c: Channel 0x7f0c2c0cc990 'SIP/zvonobot-000000bd' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000bd - start 1629282862.772741 answer 0.000000 end 1629282869.403731 dur 6.630 bill 1629282869.403 dispo NO ANSWER [Aug 18 10:34:29] DEBUG[15127] http.c: match request [ari/channels/213135] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15127] http.c: match request [ari/channels/213135] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15125] http.c: match request [ari/channels/robot_213011] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15127] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15125] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213136, detail: [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Finding handler for channels/213135 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213136': 0x7f0cb4091780 destroyed [Aug 18 10:34:29] DEBUG[15126] http.c: match request [ari/channels/213136] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Finding handler for channels/robot_213011 [Aug 18 10:34:29] DEBUG[15126] http.c: match request [ari/channels/213136] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[13682] channel.c: Channel 0x7f0c70070730 'Announcer/ARI-0000001b;1' destroying [Aug 18 10:34:29] DEBUG[15126] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Finding handler for robot_213011 [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking channels create: Didn't match robot_213011 [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15125] res_ari.c: Checking channels externalMedia: Didn't match robot_213011 [Aug 18 10:34:29] DEBUG[15125] res_ari.c: No explicit handler found for robot_213011. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Finding handler for channels/213136 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[14960] channel.c: Channel 0x7f0c880b2c10 'SIP/zvonobot-000000e0' allocated [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 From: ;tag=as3d872a68 To: ;tag=as704d0033 Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1966445502 1966445502 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15226 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as704d0033 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1966445502 1966445502 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15226 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag as704d0033 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 From: ;tag=as6f697fd0 To: ;tag=as48575d5d Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as48575d5d [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag as48575d5d [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14960] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Finding handler for 213135 [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking channels create: Didn't match 213135 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:29] DEBUG[13682] stasis.c: Destroying topic. name: cache:192/channel:1629282838.161, detail: [Aug 18 10:34:29] DEBUG[13682] stasis.c: Topic 'cache:192/channel:1629282838.161': 0x7f0c70073a00 destroyed [Aug 18 10:34:29] DEBUG[13682] stasis.c: Destroying topic. name: channel:1629282838.161, detail: [Aug 18 10:34:29] DEBUG[13682] stasis.c: Topic 'channel:1629282838.161': 0x7f0c70070650 destroyed [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Got SIP response 603 "Declined" back from 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15127] res_ari.c: Checking channels externalMedia: Didn't match 213135 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15127] res_ari.c: No explicit handler found for 213135. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68 Max-Forwards: 70 From: ;tag=as6f697fd0 To: ;tag=as48575d5d Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:29] DEBUG[14789] res_stasis.c: calls_0: Subscribing to 213173 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:29] DEBUG[14789] stasis/app.c: Channel '213173' is 1 interested in calls_0 [Aug 18 10:34:29] VERBOSE[13540] dial.c: SIP/zvonobot-00000046 is busy [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[13540] channel.c: Channel 0x7f0c9c03d0a0 'SIP/zvonobot-00000046' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #27)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6;received=159.65.48.104 From: ;tag=as4d536c24 To: ;tag=as220abe80 Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd60617" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d536c24 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as220abe80 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000046 - start 1629282836.423187 answer 0.000000 end 1629282869.450620 dur 33.027 bill 1629282869.450 dispo BUSY [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[14960] res_stasis.c: calls_0: Subscribing to 213189 [Aug 18 10:34:29] DEBUG[14960] stasis/app.c: Channel '213189' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bd60617" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 (Checking To) --From tag as4d536c24 --To-tag as220abe80 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116891@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67a05ff6 Max-Forwards: 70 From: ;tag=as4d536c24 To: Contact: Call-ID: 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 From: ;tag=as15514e30 To: ;tag=as11c4e68e Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5c816412;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as15514e30 [Aug 18 10:34:29] DEBUG[14960] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[14960] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Finding handler for 213136 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:29] DEBUG[14789] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking channels create: Didn't match 213136 [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[14789] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15126] res_ari.c: Checking channels externalMedia: Didn't match 213136 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Outgoing Call for 79821116867 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as11c4e68e [Aug 18 10:34:29] DEBUG[15016] app.c: One waitfor failed, trying another [Aug 18 10:34:29] DEBUG[15126] res_ari.c: No explicit handler found for 213136. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="121c9c5c" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 (Checking To) --From tag as15514e30 --To-tag as11c4e68e [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 4495cdde1b4fb64d03414b0e6c703d2a@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213008': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213008' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:62/channel:213008, detail: [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Outgoing Call for 79821116851 [Aug 18 10:34:29] DEBUG[14950] channel.c: Channel 0x7f0c7c0a6140 'SIP/zvonobot-000000df' allocated [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[15128] chan_sip.c: Audio is at 16168 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:62/channel:213008': 0x7f0c7c0259b0 destroyed [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213008, detail: [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14950] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] VERBOSE[15128] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213008': 0x7f0c7c016850 destroyed [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841;received=159.65.48.104 From: ;tag=as0a953bb4 To: ;tag=as0700c5a8 Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c25fd5e" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] VERBOSE[15128] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841;received=159.65.48.104 [Aug 18 10:34:29] VERBOSE[15128] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0a953bb4 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0700c5a8 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c25fd5e" [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Initializing initreq for method INVITE - callid 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116867@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 (Checking To) --From tag as0a953bb4 --To-tag as0700c5a8 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 3 [ 52]: From: ;tag=as7d725550 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 6 [ 60]: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15130] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15130] http.c: HTTP Request URI is /ari/channels/213142 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15128] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Aug 18 10:34:29] DEBUG[15128] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116892@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK08b11841 Max-Forwards: 70 From: ;tag=as0a953bb4 To: Contact: Call-ID: 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (4) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 From: ;tag=as22d5765f To: ;tag=as0550790a Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22d5765f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0550790a [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 (Checking To) --From tag as22d5765f --To-tag as0550790a [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] VERBOSE[15128] dial.c: Called zvonobot/79821116867 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213142': is 0 interested in calls_0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 From: ;tag=as1c2a52a2 To: ;tag=as025c937f Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213142' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1c2a52a2 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as025c937f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15130] http.c: match request [ari/channels/213142] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:449/channel:213142, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:449/channel:213142': 0x7f0c28104f30 destroyed [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:29] DEBUG[15130] http.c: match request [ari/channels/213142] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[14950] res_stasis.c: calls_0: Subscribing to 213186 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15130] http.c: match request [ari/channels/213142] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" [Aug 18 10:34:29] DEBUG[15130] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[14950] stasis/app.c: Channel '213186' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Finding handler for channels/213142 [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[14950] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Finding handler for 213142 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Outgoing Call for 79821116854 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 (Checking To) --From tag as1c2a52a2 --To-tag as025c937f [Aug 18 10:34:29] VERBOSE[15129] chan_sip.c: Audio is at 19388 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] DEBUG[14950] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:29] VERBOSE[15131] chan_sip.c: Audio is at 17348 [Aug 18 10:34:29] VERBOSE[15131] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking channels create: Didn't match 213142 [Aug 18 10:34:29] VERBOSE[15131] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] VERBOSE[15131] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15130] res_ari.c: Checking channels externalMedia: Didn't match 213142 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15130] res_ari.c: No explicit handler found for 213142. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213143': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[15132] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213143' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[15132] http.c: HTTP Request URI is /ari/channels/213143 [Aug 18 10:34:29] DEBUG[15132] http.c: match request [ari/channels/213143] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15132] http.c: match request [ari/channels/213143] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15132] http.c: match request [ari/channels/213143] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15132] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Finding handler for channels/213143 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:451/channel:213143, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:451/channel:213143': 0x7f0c3010d7f0 destroyed [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 From: ;tag=as11b813e8 To: ;tag=as5c9cee45 Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 652472787 652472787 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 19448 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] VERBOSE[15129] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Initializing initreq for method INVITE - callid 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as11b813e8 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5c9cee45 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116854@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] VERBOSE[15129] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] VERBOSE[15129] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 3 [ 52]: From: ;tag=as4b888052 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 6 [ 60]: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15131] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #142 [Aug 18 10:34:29] DEBUG[15131] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 652472787 652472787 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 19448 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] VERBOSE[15131] dial.c: Called zvonobot/79821116854 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 (Checking To) --From tag as11b813e8 --To-tag as5c9cee45 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '08a58ce821a53822322944fb39331df0@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Got SDP version 652472787 and unique parts [root 652472787 IN IP4 178.62.121.41] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 652472787 652472787 IN IP4 178.62.121.41... OK. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Initializing initreq for method INVITE - callid 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116851@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 3 [ 52]: From: ;tag=as5f8e1c47 [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Finding handler for 213143 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:29] DEBUG[14614] iostream.c: TCP socket error reading data: Connection reset by peer [Aug 18 10:34:29] DEBUG[14614] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213021': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking channels create: Didn't match 213143 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213021' unsubscribed from calls_0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 6 [ 60]: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:29] DEBUG[15132] res_ari.c: Checking channels externalMedia: Didn't match 213143 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:95/channel:213021, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:95/channel:213021': 0x7f0c7005bdd0 destroyed [Aug 18 10:34:29] DEBUG[15132] res_ari.c: No explicit handler found for 213143. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:29] DEBUG[15133] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:29] DEBUG[15133] http.c: HTTP Request URI is /ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/record?name=213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd&format=wav [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[15133] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/record] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15133] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/record] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[15133] http.c: match request [ari/bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/record] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15133] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Finding handler for bridges/5c24e2ba-8671-4745-b349-4500db0d3cb5/record [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Finding handler for 5c24e2ba-8671-4745-b349-4500db0d3cb5 [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15133] res_ari.c: No explicit handler found for 5c24e2ba-8671-4745-b349-4500db0d3cb5. Using wildcard bridgeId. [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Finding handler for record [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15129] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #86 [Aug 18 10:34:29] DEBUG[15129] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30074490) ICE set role failed; no ice instance [Aug 18 10:34:29] VERBOSE[15129] dial.c: Called zvonobot/79821116851 [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:29] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:29] DEBUG[15133] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213142, detail: [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30074490) RTCP setting address on RTP instance [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213142': 0x7f0c280efa80 destroyed [Aug 18 10:34:29] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0c30075fe0 -- Strict RTP learning after remote address set to: 178.62.121.41:19448 [Aug 18 10:34:29] DEBUG[15133] stasis.c: Creating topic. name: channel:1629282869.599, detail: [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:19448 [Aug 18 10:34:29] DEBUG[15133] stasis.c: Topic 'channel:1629282869.599': 0x7f0c100febb0 created [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0075ee8) from 0x7f0c147e2330 to 0x7f0c30074668 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0008b38) from 0x7f0c147e2330 to 0x7f0c30074668 [Aug 18 10:34:29] DEBUG[15133] stasis.c: Creating topic. name: cache:686/channel:1629282869.599, detail: [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb004a198) from 0x7f0c147e2330 to 0x7f0c30074668 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30074490) RTCP ignoring duplicate property [Aug 18 10:34:29] DEBUG[15133] stasis.c: Topic 'cache:686/channel:1629282869.599': 0x7f0c10072580 created [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:29] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003f setting read format path: alaw -> alaw [Aug 18 10:34:29] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000003f setting write format path: alaw -> alaw [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30074490) DTLS - ast_rtp_activate rtp=0x7f0c30075fe0 - setup and perform DTLS' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30075fe0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c30075fe0) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Strict routing enforced for session 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117008@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK337fa753 Max-Forwards: 70 From: ;tag=as11b813e8 To: ;tag=as5c9cee45 Contact: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14795] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] VERBOSE[13416] dial.c: SIP/zvonobot-0000003f answered [Aug 18 10:34:29] DEBUG[14795] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[13416] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000003f [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session timer started: 147 - 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 1768000ms [Aug 18 10:34:29] DEBUG[15135] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15135] http.c: HTTP Request URI is /ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:34:29] DEBUG[15135] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15135] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15134] http.c: HTTP opening session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487;received=159.65.48.104 From: ;tag=as123f2352 To: ;tag=as0ec69af0 Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a004025" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as123f2352 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0ec69af0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a004025" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 (Checking To) --From tag as123f2352 --To-tag as0ec69af0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116895@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK66644487 Max-Forwards: 70 From: ;tag=as123f2352 To: Contact: Call-ID: 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4560fcde Max-Forwards: 70 From: ;tag=as2c28f3a7 To: ;tag=as574e1b12 Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="26b444eb", response="d361fe0e565a2ca3b42301a6290f24e7" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4560fcde [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as2c28f3a7 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as574e1b12 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="26b444eb", response="d361fe0e565a2ca3b42301a6290f24e7" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 (Checking From) --From tag as2c28f3a7 --To-tag as574e1b12 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213139': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15135] http.c: match request [ari/bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15135] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Finding handler for bridges/95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: channel '213139' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:29] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15134] http.c: HTTP Request URI is /ari/channels/213139 [Aug 18 10:34:29] DEBUG[15134] http.c: match request [ari/channels/213139] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:456/channel:213139, detail: [Aug 18 10:34:29] DEBUG[15134] http.c: match request [ari/channels/213139] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Finding handler for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c00bc40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:29] DEBUG[15135] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:456/channel:213139': 0x7f0c200865b0 destroyed [Aug 18 10:34:29] DEBUG[15135] res_ari.c: No explicit handler found for 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060. Using wildcard bridgeId. [Aug 18 10:34:29] DEBUG[15135] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: telling all channels to leave the party [Aug 18 10:34:29] DEBUG[15134] http.c: match request [ari/channels/213139] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[13416] stasis/app.c: Channel '213032' is 2 interested in calls_0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:29] DEBUG[15135] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:29] DEBUG[15134] http.c: Match made with [ari] [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK4560fcde;received=178.62.121.41 From: ;tag=as2c28f3a7 To: ;tag=as574e1b12 Call-ID: 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15135] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: queueing action type:13 sub:1001 [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Setting 0x7f0c18091350(SIP/zvonobot-00000013) state from:0 to:1 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213143, detail: [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213143': 0x7f0c30030ac0 destroyed [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213021, detail: [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling stasis bridge destructor [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology stop [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge 95aa254a-8cb0-4e7f-94b3-e5d21f2bb060: calling simple_bridge technology destructor [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session timer stopped: 59 - 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pulling 0x7f0c18091350(SIP/zvonobot-00000013) [Aug 18 10:34:29] VERBOSE[13704] bridge_channel.c: Channel SIP/zvonobot-00000013 left 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213021': 0x7f0c7005b360 destroyed [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c18091350(SIP/zvonobot-00000013) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Setting 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) state from:0 to:2 [Aug 18 10:34:29] DEBUG[15135] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Finding handler for channels/213139 [Aug 18 10:34:29] DEBUG[15135] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[14955] channel.c: Channel 0x7f0c7803e510 'SIP/zvonobot-000000e2' allocated [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14955] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[15136] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Finding handler for 213139 [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking channels create: Didn't match 213139 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: channel:213139, detail: [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'channel:213139': 0x7f0c2007dc20 destroyed [Aug 18 10:34:29] DEBUG[15134] res_ari.c: Checking channels externalMedia: Didn't match 213139 [Aug 18 10:34:29] DEBUG[15134] res_ari.c: No explicit handler found for 213139. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[15136] http.c: HTTP Request URI is /ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000013 - start 1629282824.240569 answer 1629282839.719498 end 1629282869.576825 dur 45.336 bill 29.857 dispo ANSWERED [Aug 18 10:34:29] DEBUG[13704] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[13704] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[13704] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13704] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13704] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13704] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677 is already using the new technology. [Aug 18 10:34:29] DEBUG[14937] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: pulling 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) [Aug 18 10:34:29] VERBOSE[14937] bridge_channel.c: Channel Announcer/ARI-00000049;2 left 'simple_bridge' stasis-bridge <45640e14-e267-477d-81ea-fbac374f9677> [Aug 18 10:34:29] DEBUG[14937] bridge_channel.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677: 0x7f0c1c12e5c0(Announcer/ARI-00000049;2) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Bridge is returning 0x7f0c18091350(SIP/zvonobot-00000013) to read format alaw [Aug 18 10:34:29] DEBUG[15136] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[13704] channel.c: Channel SIP/zvonobot-00000013 setting read format path: ulaw -> alaw [Aug 18 10:34:29] DEBUG[13704] bridge_channel.c: Bridge is returning 0x7f0c18091350(SIP/zvonobot-00000013) to write format alaw [Aug 18 10:34:29] DEBUG[14937] bridge_native_rtp.c: Bridge '45640e14-e267-477d-81ea-fbac374f9677' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[14937] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[14937] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14937] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14937] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[14937] bridge.c: Bridge 45640e14-e267-477d-81ea-fbac374f9677 is already using the new technology. [Aug 18 10:34:29] DEBUG[13704] channel.c: Channel SIP/zvonobot-00000013 setting write format path: alaw -> ulaw [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:29] DEBUG[13704] stasis/control.c: 212982, 45640e14-e267-477d-81ea-fbac374f9677: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13704] stasis/app.c: bridge '45640e14-e267-477d-81ea-fbac374f9677': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[13704] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[14937] channel.c: Channel 0x7f0c1c157030 'Announcer/ARI-00000049;2' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[12971] stasis/control.c: 212982: Channel departing bridge [Aug 18 10:34:29] DEBUG[12971] bridge.c: Waiting for 0x7f0c18091350(SIP/zvonobot-00000013) bridge thread to die. [Aug 18 10:34:29] DEBUG[12971] stasis/app.c: channel '212982': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[12971] channel.c: Channel 0x7f0c1c026d20 'SIP/zvonobot-00000013' hanging up. Refs: 3 [Aug 18 10:34:29] DEBUG[20535] devicestate.c: Changing state for Recorder/ARI - state 2 (In use) [Aug 18 10:34:29] DEBUG[15136] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[13638] channel.c: Channel 0x7f0c800507f0 'Recorder/ARI-0000001a;1' destroying [Aug 18 10:34:29] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for UnicastRTP - 127.0.0.1:50194 [Aug 18 10:34:29] DEBUG[20616] app_queue.c: Device 'Recorder/ARI' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 18 10:34:29] DEBUG[14957] channel.c: Channel 0x7f0c80038850 'SIP/zvonobot-000000e1' allocated [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74;received=159.65.48.104 From: ;tag=as5e0e197a To: ;tag=as4d1db78a Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10b3148a" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[15136] http.c: match request [ari/bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15136] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[14957] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7ed49d74;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5e0e197a [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Finding handler for bridges/e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:29] DEBUG[13638] stasis.c: Destroying topic. name: cache:184/channel:1629282837.155, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4d1db78a [Aug 18 10:34:29] DEBUG[13638] stasis.c: Topic 'cache:184/channel:1629282837.155': 0x7f0c80052f90 destroyed [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[13638] stasis.c: Destroying topic. name: channel:1629282837.155, detail: [Aug 18 10:34:29] DEBUG[13638] stasis.c: Topic 'channel:1629282837.155': 0x7f0c80052540 destroyed [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[14955] res_stasis.c: calls_0: Subscribing to 213187 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10b3148a" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 (Checking To) --From tag as5e0e197a --To-tag as4d1db78a [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Finding handler for e0573cd4-75f6-4425-a1e4-83029f01aa9a [Aug 18 10:34:29] DEBUG[15136] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15136] res_ari.c: No explicit handler found for e0573cd4-75f6-4425-a1e4-83029f01aa9a. Using wildcard bridgeId. [Aug 18 10:34:29] DEBUG[15136] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: telling all channels to leave the party [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1ac623ca476f0eb072a1424919136fca@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14955] stasis/app.c: Channel '213187' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[14955] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[14957] res_stasis.c: calls_0: Subscribing to 213188 [Aug 18 10:34:29] DEBUG[14957] stasis/app.c: Channel '213188' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Outgoing Call for 79821116853 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[14957] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[14957] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[14955] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:29] DEBUG[15136] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:29] DEBUG[14722] channel.c: Channel 0x7f0c3c03b3f0 'Announcer/ARI-0000003a;2' destroying [Aug 18 10:34:29] DEBUG[15136] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: queueing action type:13 sub:1001 [Aug 18 10:34:29] DEBUG[15136] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:29] DEBUG[13330] app.c: One waitfor failed, trying another [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[14653] bridge_channel.c: Setting 0x7f0c3c013500(Recorder/ARI-00000035;2) state from:0 to:1 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: calling stasis bridge destructor [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: calling simple_bridge technology stop [Aug 18 10:34:29] DEBUG[20534] bridge.c: Bridge e0573cd4-75f6-4425-a1e4-83029f01aa9a: calling simple_bridge technology destructor [Aug 18 10:34:29] DEBUG[14665] channel.c: Channel 0x7f0c3c12ca30 'Recorder/ARI-00000035;1' destroying [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15136] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14722] stasis.c: Destroying topic. name: cache:465/channel:1629282857.404, detail: [Aug 18 10:34:29] DEBUG[14722] stasis.c: Topic 'cache:465/channel:1629282857.404': 0x7f0c3c03d5b0 destroyed [Aug 18 10:34:29] DEBUG[14722] stasis.c: Destroying topic. name: channel:1629282857.404, detail: [Aug 18 10:34:29] DEBUG[14722] stasis.c: Topic 'channel:1629282857.404': 0x7f0c3c03cbc0 destroyed [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Outgoing Call for 79821116852 [Aug 18 10:34:29] VERBOSE[15139] chan_sip.c: Audio is at 12262 [Aug 18 10:34:29] VERBOSE[15139] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[14943] channel.c: Channel 0x7f0c400a9b80 'SIP/zvonobot-000000e3' allocated [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] VERBOSE[15139] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] VERBOSE[15139] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14665] stasis.c: Destroying topic. name: cache:404/channel:1629282853.345, detail: [Aug 18 10:34:29] DEBUG[14665] stasis.c: Topic 'cache:404/channel:1629282853.345': 0x7f0c3c004d90 destroyed [Aug 18 10:34:29] DEBUG[14665] stasis.c: Destroying topic. name: channel:1629282853.345, detail: [Aug 18 10:34:29] DEBUG[14665] stasis.c: Topic 'channel:1629282853.345': 0x7f0c3c119170 destroyed [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14943] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[14653] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: pulling 0x7f0c3c013500(Recorder/ARI-00000035;2) [Aug 18 10:34:29] VERBOSE[14653] bridge_channel.c: Channel Recorder/ARI-00000035;2 left 'simple_bridge' stasis-bridge <21515bb0-91f2-4ad5-852f-8721c870cad7> [Aug 18 10:34:29] DEBUG[14653] bridge_channel.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7: 0x7f0c3c013500(Recorder/ARI-00000035;2) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[15140] chan_sip.c: Audio is at 11578 [Aug 18 10:34:29] VERBOSE[15140] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[14943] res_stasis.c: calls_0: Subscribing to 213185 [Aug 18 10:34:29] VERBOSE[15140] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[14943] stasis/app.c: Channel '213185' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[14653] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[14653] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[14653] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14653] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[14653] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[14653] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7 is already using the new technology. [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Outgoing Call for 79821116855 [Aug 18 10:34:29] DEBUG[14943] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] VERBOSE[15140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[14943] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[14653] bridge_channel.c: Bridge is returning 0x7f0c3c013500(Recorder/ARI-00000035;2) to write format slin [Aug 18 10:34:29] DEBUG[14653] channel.c: Channel Recorder/ARI-00000035;2 setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[14653] channel.c: Channel 0x7f0c3c149bd0 'Recorder/ARI-00000035;2' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09;received=159.65.48.104 From: ;tag=as41f91965 To: ;tag=as120618a4 Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08d0058d" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31c4ad09;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as41f91965 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as120618a4 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08d0058d" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[15142] chan_sip.c: Audio is at 14186 [Aug 18 10:34:29] VERBOSE[15142] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Initializing initreq for method INVITE - callid 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 (Checking To) --From tag as41f91965 --To-tag as120618a4 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116853@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5bdcfa8b537506523572ed737edb75f5@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #142 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 3 [ 52]: From: ;tag=as6ae39521 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #142)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 6 [ 60]: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15139] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #153 [Aug 18 10:34:29] DEBUG[15139] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #27)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[15139] dial.c: Called zvonobot/79821116853 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] VERBOSE[15142] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] VERBOSE[15142] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Initializing initreq for method INVITE - callid 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116852@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 From: ;tag=as54e004b1 To: ;tag=as58786f9e Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1451248754 1451248754 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 17184 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Initializing initreq for method INVITE - callid 21405c0201d61407119338763dc16673@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5bf8c9b8;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116855@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as54e004b1 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as58786f9e [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 3 [ 52]: From: ;tag=as76c090e3 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 3 [ 52]: From: ;tag=as6137e79e [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 6 [ 60]: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 6 [ 60]: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15142] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #155 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[15142] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1451248754 1451248754 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15140] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 17184 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #157 [Aug 18 10:34:29] DEBUG[15140] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[15142] dial.c: Called zvonobot/79821116855 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] VERBOSE[15140] dial.c: Called zvonobot/79821116852 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:29] DEBUG[14796] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14796] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 (Checking To) --From tag as54e004b1 --To-tag as58786f9e [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Strict routing enforced for session 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117030@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK389026d4 Max-Forwards: 70 From: ;tag=as54e004b1 To: ;tag=as58786f9e Contact: Call-ID: 56adfdbd34ef766b7a7d7a74079c107b@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14798] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[14798] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[14801] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14801] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8;received=159.65.48.104 From: ;tag=as7a3cd0ea To: ;tag=as3e98a678 Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4676557d" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7a3cd0ea [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3e98a678 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4676557d" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 (Checking To) --From tag as7a3cd0ea --To-tag as3e98a678 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786;received=159.65.48.104 From: ;tag=as6230d06d To: ;tag=as73bad0b4 Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="14c94304" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d101786;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6230d06d [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as73bad0b4 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="14c94304" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 (Checking To) --From tag as6230d06d --To-tag as73bad0b4 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 0e7a32cf382bc2517c20e72c652f2ed0@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0;received=159.65.48.104 From: ;tag=as2e1ef431 To: ;tag=as14121ff0 Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b70e34e" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4edad0a0;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2e1ef431 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as14121ff0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b70e34e" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 (Checking To) --From tag as2e1ef431 --To-tag as14121ff0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2a1fc5905a1e93bd0ec28a42030d0412@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[14797] channel.c: Channel 0x7f0c8408ee40 'Announcer/ARI-0000004a;2' allocated [Aug 18 10:34:29] DEBUG[14797] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:29] DEBUG[14797] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000004a;1' [Aug 18 10:34:29] DEBUG[15147] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c84119870(Announcer/ARI-0000004a;2) is joining [Aug 18 10:34:29] DEBUG[14804] channel.c: Channel 0x7f0c9c086b00 'Recorder/ARI-0000004b;2' allocated [Aug 18 10:34:29] DEBUG[14804] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:29] DEBUG[14800] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14800] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15148] bridge_channel.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1: 0x7f0c9c08ea10(Recorder/ARI-0000004b;2) is joining [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 From: ;tag=as0e0b214d To: ;tag=as0175bfce Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1568239288 1568239288 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 13490 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:29] DEBUG[15147] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pushing 0x7f0c84119870(Announcer/ARI-0000004a;2) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK130dd081;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[15147] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0e0b214d [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0175bfce [Aug 18 10:34:29] VERBOSE[15147] bridge_channel.c: Channel Announcer/ARI-0000004a;2 joined 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15148] bridge_channel.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1: pushing 0x7f0c9c08ea10(Recorder/ARI-0000004b;2) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:29] DEBUG[14962] channel.c: Channel 0x7f0c940cad30 'SIP/zvonobot-000000e4' allocated [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1568239288 1568239288 IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 13490 RTP/AVP 0 8 101 [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14962] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[15147] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[15147] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15147] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15147] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15147] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[15147] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:29] DEBUG[15147] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c84119870(Announcer/ARI-0000004a;2) is joining simple_bridge technology [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:29] DEBUG[14797] res_stasis_playback.c: 1629282862.488: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:29] DEBUG[14809] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14809] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[13698] app.c: One waitfor failed, trying another [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:29] DEBUG[14797] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:29] DEBUG[14797] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15151] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:29] WARNING[14882] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-0000003f;1 [Aug 18 10:34:29] DEBUG[15148] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:29] VERBOSE[15148] bridge_channel.c: Channel Recorder/ARI-0000004b;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[15151] http.c: HTTP Request URI is /ari/channels/1629282833.102 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:29] DEBUG[15151] http.c: match request [ari/channels/1629282833.102] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15151] http.c: match request [ari/channels/1629282833.102] with handler [phoneprov] len 9 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:29] DEBUG[15151] http.c: match request [ari/channels/1629282833.102] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15151] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Finding handler for channels/1629282833.102 [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Finding handler for 1629282833.102 [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking channels create: Didn't match 1629282833.102 [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 (Checking To) --From tag as0e0b214d --To-tag as0175bfce [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Stopping retransmission on '2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Got SDP version 1568239288 and unique parts [root 1568239288 IN IP4 178.62.121.41] [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 1568239288 1568239288 IN IP4 178.62.121.41... OK. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c36530) ICE set role failed; no ice instance [Aug 18 10:34:29] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c36530) RTCP setting address on RTP instance [Aug 18 10:34:29] VERBOSE[20585] res_rtp_asterisk.c: 0x2c38080 -- Strict RTP learning after remote address set to: 178.62.121.41:13490 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:13490 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb00157b8) from 0x7f0c147e2330 to 0x2c36708 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0127208) from 0x7f0c147e2330 to 0x2c36708 [Aug 18 10:34:29] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb003fb58) from 0x7f0c147e2330 to 0x2c36708 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c36530) RTCP ignoring duplicate property [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:29] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000004d setting read format path: alaw -> alaw [Aug 18 10:34:29] DEBUG[20585] channel.c: Channel SIP/zvonobot-0000004d setting write format path: alaw -> alaw [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c36530) DTLS - ast_rtp_activate rtp=0x2c38080 - setup and perform DTLS' [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c38080) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:29] DEBUG[20585] res_rtp_asterisk.c: (0x2c38080) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:29] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:29] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:29] DEBUG[15151] res_ari.c: Checking channels externalMedia: Didn't match 1629282833.102 [Aug 18 10:34:29] DEBUG[15151] res_ari.c: No explicit handler found for 1629282833.102. Using wildcard channelId. [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15152] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15150] channel.c: Channel Announcer/ARI-0000004a;1 setting write format path: gsm -> slin [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116997@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c75ec84 Max-Forwards: 70 From: ;tag=as0e0b214d To: ;tag=as0175bfce Contact: Call-ID: 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15150] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:29] VERBOSE[15150] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:29] DEBUG[15148] bridge_native_rtp.c: Bridge 'c8381fea-1239-48c9-a6e3-1d9ad7226cf1'. Checking compatability for channels 'SIP/zvonobot-0000000f' and 'Recorder/ARI-0000004b;2' [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Session timer started: 145 - 2e9c69844da5e0bc1a22c1073d3b75bf@159.65.48.104:5060 1768000ms [Aug 18 10:34:29] DEBUG[15152] http.c: HTTP Request URI is /ari/playbacks/fe7814b7-d6cc-4637-8d66-7450ef22c828 [Aug 18 10:34:29] DEBUG[15148] bridge_native_rtp.c: Bridge 'c8381fea-1239-48c9-a6e3-1d9ad7226cf1' can not use native RTP bridge as could not get details [Aug 18 10:34:29] DEBUG[15148] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15148] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15152] http.c: match request [ari/playbacks/fe7814b7-d6cc-4637-8d66-7450ef22c828] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15148] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15148] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] VERBOSE[13568] dial.c: SIP/zvonobot-0000004d answered [Aug 18 10:34:29] VERBOSE[13568] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-0000004d [Aug 18 10:34:29] DEBUG[15152] http.c: match request [ari/playbacks/fe7814b7-d6cc-4637-8d66-7450ef22c828] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[14623] iostream.c: TCP socket error reading data: Connection reset by peer [Aug 18 10:34:29] DEBUG[14623] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15148] bridge.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1 is already using the new technology. [Aug 18 10:34:29] DEBUG[15153] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[13568] stasis/app.c: Channel '213043' is 2 interested in calls_0 [Aug 18 10:34:29] DEBUG[15148] bridge.c: Bridge c8381fea-1239-48c9-a6e3-1d9ad7226cf1: 0x7f0c9c08ea10(Recorder/ARI-0000004b;2) is joining simple_bridge technology [Aug 18 10:34:29] DEBUG[14962] res_stasis.c: calls_0: Subscribing to 213193 [Aug 18 10:34:29] DEBUG[14962] stasis/app.c: Channel '213193' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel Recorder/ARI-0000004b;2 setting read format path: slin -> slin [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel SIP/zvonobot-0000000f setting write format path: slin -> ulaw [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel SIP/zvonobot-0000000f setting read format path: ulaw -> slin [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel Recorder/ARI-0000004b;2 setting write format path: slin -> slin [Aug 18 10:34:29] DEBUG[14962] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[14962] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15152] http.c: match request [ari/playbacks/fe7814b7-d6cc-4637-8d66-7450ef22c828] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[14808] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Outgoing Call for 79821116847 [Aug 18 10:34:29] DEBUG[15152] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[14808] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd;received=159.65.48.104 From: ;tag=as3f0bc324 To: ;tag=as797a96ff Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5358a1b7" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f0bc324 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as797a96ff [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Finding handler for playbacks/fe7814b7-d6cc-4637-8d66-7450ef22c828 [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Finding handler for playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5358a1b7" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15153] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212967&app=calls_0&format=slin16&external_host=127.0.0.1%3A50044 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 (Checking To) --From tag as3f0bc324 --To-tag as797a96ff [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #153 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #153)) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #155 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[14804] res_stasis_recording.c: 1629282863.492: Sending record(212980_dquqCSdGIzjcjMVjdrEqzTurcKoxWjar.wav) command [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #155)) [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[14804] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #157 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:29] DEBUG[14804] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel Recorder/ARI-0000004b;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:29] DEBUG[15148] channel.c: Channel Recorder/ARI-0000004b;2 setting write format path: alaw -> slin [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:29] DEBUG[15153] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #157)) [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Finding handler for fe7814b7-d6cc-4637-8d66-7450ef22c828 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15152] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15152] res_ari.c: No explicit handler found for fe7814b7-d6cc-4637-8d66-7450ef22c828. Using wildcard playbackId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] DEBUG[15150] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:29] DEBUG[15150] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:29] DEBUG[15150] channel.c: Channel Announcer/ARI-0000004a;1 setting write format path: slin -> slin [Aug 18 10:34:29] NOTICE[15150] res_stasis_playback.c: 1629282862.488: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:29] DEBUG[15150] channel.c: Channel 0x7f0c84148b70 'Announcer/ARI-0000004a;1' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15153] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15153] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15153] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:29] DEBUG[15156] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15152] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] VERBOSE[15154] chan_sip.c: Audio is at 13644 [Aug 18 10:34:29] DEBUG[15152] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15155] app.c: play_and_record: , /var/spool/asterisk/recording/212980_dquqCSdGIzjcjMVjdrEqzTurcKoxWjar, 'wav' [Aug 18 10:34:29] DEBUG[15156] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:29] DEBUG[15156] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15155] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Finding handler for channels [Aug 18 10:34:29] VERBOSE[15154] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15153] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:29] DEBUG[15153] netsock2.c: Splitting '127.0.0.1:50044' into... [Aug 18 10:34:29] DEBUG[15153] netsock2.c: ...host '127.0.0.1' and port '50044'. [Aug 18 10:34:29] DEBUG[15153] netsock2.c: Splitting '127.0.0.1:50044' into... [Aug 18 10:34:29] DEBUG[15153] netsock2.c: ...host '127.0.0.1' and port '50044'. [Aug 18 10:34:29] DEBUG[15153] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:29] DEBUG[15153] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c400c76f0' [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) RTP allocated port 17026 [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE creating session 127.0.0.1:17026 (17026) [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE create [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE add system candidates [Aug 18 10:34:29] DEBUG[15153] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:29] DEBUG[15153] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:29] DEBUG[15156] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:29] VERBOSE[15155] app.c: x=0, open writing: /var/spool/asterisk/recording/212980_dquqCSdGIzjcjMVjdrEqzTurcKoxWjar format: wav, 0x7f0c700913f0 [Aug 18 10:34:29] DEBUG[15158] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15156] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15156] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE add candidate: 159.65.48.104:17026, 2130706431 [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[15153] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:29] DEBUG[15153] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:29] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE add candidate: 10.131.0.10:17026, 2130706431 [Aug 18 10:34:29] DEBUG[15153] rtp_engine.c: RTP instance '0x7f0c400c76f0' is setup and ready to go [Aug 18 10:34:29] DEBUG[15153] stasis.c: Creating topic. name: channel:robot_212967, detail: [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[15157] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:29] DEBUG[15153] stasis.c: Topic is already exist. name: channel:robot_212967, detail: [Aug 18 10:34:29] VERBOSE[15154] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] VERBOSE[15154] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] DEBUG[15156] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15153] stasis.c: Creating topic. name: cache:687/channel:robot_212967, detail: [Aug 18 10:34:29] DEBUG[15153] stasis.c: Topic 'cache:687/channel:robot_212967': 0x7f0c40067b90 created [Aug 18 10:34:29] DEBUG[15156] stasis.c: Creating topic. name: bridge:caa6f495-4822-4780-8a7f-595474d7d555, detail: [Aug 18 10:34:29] DEBUG[15156] stasis.c: Topic 'bridge:caa6f495-4822-4780-8a7f-595474d7d555': 0x7f0c7c016630 created [Aug 18 10:34:29] DEBUG[15156] stasis.c: Creating topic. name: cache:688/bridge:caa6f495-4822-4780-8a7f-595474d7d555, detail: [Aug 18 10:34:29] DEBUG[15156] stasis.c: Topic 'cache:688/bridge:caa6f495-4822-4780-8a7f-595474d7d555': 0x7f0c7c025d20 created [Aug 18 10:34:29] DEBUG[15156] bridge_native_rtp.c: Bridge 'caa6f495-4822-4780-8a7f-595474d7d555' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[15156] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15156] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15156] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:29] DEBUG[15156] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[15156] bridge.c: Bridge caa6f495-4822-4780-8a7f-595474d7d555: calling simple_bridge technology constructor [Aug 18 10:34:29] DEBUG[15156] bridge.c: Bridge caa6f495-4822-4780-8a7f-595474d7d555: calling simple_bridge technology start [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 From: ;tag=as3056f2e0 To: ;tag=as20b331c1 Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15156] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15158] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15158] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15158] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15158] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15156] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15159] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060': is 0 interested in calls_0 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Initializing initreq for method INVITE - callid 5676735873320902534290d27970f7c7@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[15157] http.c: HTTP Request URI is /ari/channels/robot_212991 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3056f2e0 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116847@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15158] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15158] stasis.c: Creating topic. name: bridge:ef2f0ed6-7b56-46c3-a894-dc0114b2fb34, detail: [Aug 18 10:34:29] DEBUG[15158] stasis.c: Topic 'bridge:ef2f0ed6-7b56-46c3-a894-dc0114b2fb34': 0x7f0c84049da0 created [Aug 18 10:34:29] DEBUG[15158] stasis.c: Creating topic. name: cache:689/bridge:ef2f0ed6-7b56-46c3-a894-dc0114b2fb34, detail: [Aug 18 10:34:29] DEBUG[15159] http.c: HTTP Request URI is /ari/channels/212980/snoop?app=calls_0&spy=in [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: bridge '95aa254a-8cb0-4e7f-94b3-e5d21f2bb060' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:181/bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:181/bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060': 0x7f0c2406c820 destroyed [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'bridge:95aa254a-8cb0-4e7f-94b3-e5d21f2bb060': 0x7f0c240f2ab0 destroyed [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:29] DEBUG[15160] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15160] http.c: HTTP Request URI is /ari/playbacks/87067a7f-a8f0-4dcb-9b26-73f965028d4f [Aug 18 10:34:29] DEBUG[15160] http.c: match request [ari/playbacks/87067a7f-a8f0-4dcb-9b26-73f965028d4f] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15160] http.c: match request [ari/playbacks/87067a7f-a8f0-4dcb-9b26-73f965028d4f] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15160] http.c: match request [ari/playbacks/87067a7f-a8f0-4dcb-9b26-73f965028d4f] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15160] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20b331c1 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[15158] stasis.c: Topic 'cache:689/bridge:ef2f0ed6-7b56-46c3-a894-dc0114b2fb34': 0x7f0c8408bbd0 created [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 3 [ 52]: From: ;tag=as3fd361cb [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15159] http.c: match request [ari/channels/212980/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15159] http.c: match request [ari/channels/212980/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 6 [ 60]: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15159] http.c: match request [ari/channels/212980/snoop] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 (Checking To) --From tag as3056f2e0 --To-tag as20b331c1 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis/app.c: bridge 'e0573cd4-75f6-4425-a1e4-83029f01aa9a' unsubscribed from calls_0 [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: cache:203/bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'cache:203/bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a': 0x7f0c980a77c0 destroyed [Aug 18 10:34:29] DEBUG[20620] stasis.c: Destroying topic. name: bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a, detail: [Aug 18 10:34:29] DEBUG[20620] stasis.c: Topic 'bridge:e0573cd4-75f6-4425-a1e4-83029f01aa9a': 0x7f0c980a8850 destroyed [Aug 18 10:34:29] DEBUG[15159] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Finding handler for channels/212980/snoop [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #142 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #142)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15158] bridge_native_rtp.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[15158] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15158] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15158] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:29] DEBUG[15158] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[15158] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: calling simple_bridge technology constructor [Aug 18 10:34:29] DEBUG[15158] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: calling simple_bridge technology start [Aug 18 10:34:29] DEBUG[15161] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15161] http.c: HTTP Request URI is /ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/addChannel?channel=213032 [Aug 18 10:34:29] DEBUG[15161] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15161] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15161] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/addChannel] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15161] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Finding handler for bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/addChannel [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Finding handler for ef2f0ed6-7b56-46c3-a894-dc0114b2fb34 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15157] http.c: match request [ari/channels/robot_212991] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15158] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[15158] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15157] http.c: match request [ari/channels/robot_212991] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15157] http.c: match request [ari/channels/robot_212991] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15157] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] VERBOSE[15154] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #146 [Aug 18 10:34:29] DEBUG[15154] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Finding handler for playbacks/87067a7f-a8f0-4dcb-9b26-73f965028d4f [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Finding handler for 212980 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channels create: Didn't match 212980 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channels externalMedia: Didn't match 212980 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: No explicit handler found for 212980. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Finding handler for snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Finding handler for channels/robot_212991 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Finding handler for playbacks [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[14685] res_rtp_asterisk.c: (0x7f0ca40ff9e0) RTCP got report of 76 bytes from 178.62.121.41:15869 [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:29] DEBUG[15159] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594;received=159.65.48.104 From: ;tag=as6d27c109 To: ;tag=as276fddfb Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63acfefa" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6985b594;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6d27c109 [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as276fddfb [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:29] DEBUG[14965] channel.c: Channel 0x7f0c9c0e46a0 'SIP/zvonobot-000000e5' allocated [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:29] DEBUG[14965] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[14965] res_stasis.c: calls_0: Subscribing to 213191 [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Finding handler for robot_212991 [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking channels create: Didn't match robot_212991 [Aug 18 10:34:29] DEBUG[14965] stasis/app.c: Channel '213191' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[14965] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] VERBOSE[15154] dial.c: Called zvonobot/79821116847 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63acfefa" [Aug 18 10:34:29] DEBUG[14965] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15161] res_ari.c: No explicit handler found for ef2f0ed6-7b56-46c3-a894-dc0114b2fb34. Using wildcard bridgeId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 (Checking To) --From tag as6d27c109 --To-tag as276fddfb [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 70e98101438712b7436e96ce41ef49c2@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6282ms with no response [Aug 18 10:34:29] WARNING[20585] chan_sip.c: Hanging up call 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[14821] channel.c: Channel 0x7f0cac056530 'SIP/zvonobot-000000a6' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000a6 - start 1629282863.416655 answer 0.000000 end 1629282869.869016 dur 6.452 bill 1629282869.869 dispo NO ANSWER [Aug 18 10:34:29] DEBUG[15157] res_ari.c: Checking channels externalMedia: Didn't match robot_212991 [Aug 18 10:34:29] DEBUG[15157] res_ari.c: No explicit handler found for robot_212991. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Outgoing Call for 79821116849 [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Finding handler for 87067a7f-a8f0-4dcb-9b26-73f965028d4f [Aug 18 10:34:29] DEBUG[15160] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:29] DEBUG[15160] res_ari.c: No explicit handler found for 87067a7f-a8f0-4dcb-9b26-73f965028d4f. Using wildcard playbackId. [Aug 18 10:34:29] DEBUG[15160] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[14946] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:29] DEBUG[14946] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:29] DEBUG[14946] channel.c: Channel Announcer/ARI-00000049;1 setting write format path: slin -> slin [Aug 18 10:34:29] NOTICE[14946] res_stasis_playback.c: 1629282862.480: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:29] DEBUG[15160] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[15165] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15165] http.c: HTTP Request URI is /ari/channels/robot_212982 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:29] VERBOSE[15163] chan_sip.c: Audio is at 18212 [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Finding handler for addChannel [Aug 18 10:34:29] DEBUG[15165] http.c: match request [ari/channels/robot_212982] with handler [httpstatus] len 10 [Aug 18 10:34:29] VERBOSE[15163] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:29] VERBOSE[15163] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[15165] http.c: match request [ari/channels/robot_212982] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[15165] http.c: match request [ari/channels/robot_212982] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15165] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Finding handler for channels/robot_212982 [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[14946] channel.c: Channel 0x7f0c1c0366a0 'Announcer/ARI-00000049;1' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Finding handler for robot_212982 [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking channels create: Didn't match robot_212982 [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15165] res_ari.c: Checking channels externalMedia: Didn't match robot_212982 [Aug 18 10:34:29] VERBOSE[15163] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 From: ;tag=as64e6e544 To: ;tag=as189a4383 Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[15165] res_ari.c: No explicit handler found for robot_212982. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Initializing initreq for method INVITE - callid 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116849@178.62.121.41 SIP/2.0 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 3 [ 52]: From: ;tag=as1781187a [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:29] DEBUG[14678] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 6 [ 60]: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:29 GMT [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:29] DEBUG[14678] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[14991] channel.c: Channel 0x7f0c1c048410 'Announcer/ARI-0000004f;1' allocated [Aug 18 10:34:29] DEBUG[14995] channel.c: Channel 0x7f0c1805f5c0 'Announcer/ARI-0000004e;1' allocated [Aug 18 10:34:29] DEBUG[14995] stasis.c: Creating topic. name: channel:1629282869.602, detail: [Aug 18 10:34:29] DEBUG[14995] stasis.c: Topic 'channel:1629282869.602': 0x7f0c1803f5b0 created [Aug 18 10:34:29] DEBUG[14995] stasis.c: Creating topic. name: cache:690/channel:1629282869.602, detail: [Aug 18 10:34:29] DEBUG[14991] stasis.c: Creating topic. name: channel:1629282869.601, detail: [Aug 18 10:34:29] DEBUG[15161] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:29] VERBOSE[15163] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #59 [Aug 18 10:34:29] DEBUG[15163] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[14991] stasis.c: Topic 'channel:1629282869.601': 0x7f0c1c017be0 created [Aug 18 10:34:29] DEBUG[14991] stasis.c: Creating topic. name: cache:691/channel:1629282869.601, detail: [Aug 18 10:34:29] DEBUG[14995] stasis.c: Topic 'cache:690/channel:1629282869.602': 0x7f0c180c8ea0 created [Aug 18 10:34:29] DEBUG[14991] stasis.c: Topic 'cache:691/channel:1629282869.601': 0x7f0c1c04ca70 created [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK292a286f;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as64e6e544 [Aug 18 10:34:29] VERBOSE[15163] dial.c: Called zvonobot/79821116849 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as189a4383 [Aug 18 10:34:29] DEBUG[15161] stasis/control.c: 213032: Sending channel add_to_bridge command [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c652001" [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 (Checking To) --From tag as64e6e544 --To-tag as189a4383 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:29] DEBUG[15167] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15167] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:29] DEBUG[15167] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15167] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[15167] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15167] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[14828] stasis.c: Creating topic. name: channel:1629282869.603, detail: [Aug 18 10:34:29] DEBUG[14828] stasis.c: Topic 'channel:1629282869.603': 0x7f0c380817c0 created [Aug 18 10:34:29] DEBUG[14828] stasis.c: Creating topic. name: cache:692/channel:1629282869.603, detail: [Aug 18 10:34:29] DEBUG[14828] stasis.c: Topic 'cache:692/channel:1629282869.603': 0x7f0c38048f70 created [Aug 18 10:34:29] DEBUG[14820] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14820] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[14695] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00' [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 62fe87cb521a0f054122f221360b89e6@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15168] http.c: HTTP opening session. Top level [Aug 18 10:34:29] DEBUG[15168] http.c: HTTP Request URI is /ari/channels/213007 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (4) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:29] DEBUG[13947] bridge_channel.c: Setting 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) state from:0 to:1 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Finding handler for bridges [Aug 18 10:34:29] DEBUG[14695] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:29] DEBUG[15168] http.c: match request [ari/channels/213007] with handler [httpstatus] len 10 [Aug 18 10:34:29] DEBUG[15168] http.c: match request [ari/channels/213007] with handler [phoneprov] len 9 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15168] http.c: match request [ari/channels/213007] with handler [ari] len 3 [Aug 18 10:34:29] DEBUG[15168] http.c: Match made with [ari] [Aug 18 10:34:29] DEBUG[14695] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #146 (1) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #146)) [Aug 18 10:34:29] DEBUG[13947] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: pulling 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] VERBOSE[13947] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 left 'simple_bridge' stasis-bridge [Aug 18 10:34:29] DEBUG[13947] bridge_channel.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f: 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) is leaving simple_bridge technology [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #153 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #153)) [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[13947] bridge_native_rtp.c: Bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f' can not use native RTP bridge as two channels are required [Aug 18 10:34:29] DEBUG[13947] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Finding handler for channels/213007 [Aug 18 10:34:29] DEBUG[13947] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Finding handler for channels [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:29] DEBUG[13947] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:29] DEBUG[13947] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:29] DEBUG[13416] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000003f [Aug 18 10:34:29] DEBUG[13947] bridge.c: Bridge beb17a84-adfc-4fa3-b7a8-31977a540c1f is already using the new technology. [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:29] DEBUG[13416] stasis/control.c: 213032: Adding to bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34 [Aug 18 10:34:29] DEBUG[13416] stasis/app.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34' is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:29] DEBUG[13947] bridge_channel.c: Bridge is returning 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) to write format slin16 [Aug 18 10:34:29] DEBUG[15167] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:29] DEBUG[13947] channel.c: Channel UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00 setting write format path: slin16 -> slin16 [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:29] DEBUG[15167] stasis.c: Creating topic. name: bridge:8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95, detail: [Aug 18 10:34:29] DEBUG[15167] stasis.c: Topic 'bridge:8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95': 0x7f0ca801d200 created [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc;received=159.65.48.104 From: ;tag=as40bb47c8 To: ;tag=as1e663664 Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00ff45a1" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK096f61bc;received=159.65.48.104 [Aug 18 10:34:29] DEBUG[14812] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as40bb47c8 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1e663664 [Aug 18 10:34:29] DEBUG[14812] http.c: HTTP closing session. Top level [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Finding handler for 213007 [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking channels create: Didn't match 213007 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00ff45a1" [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:29] DEBUG[15168] res_ari.c: Checking channels externalMedia: Didn't match 213007 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:29] DEBUG[15168] res_ari.c: No explicit handler found for 213007. Using wildcard channelId. [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: = Looking for Call ID: 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 (Checking To) --From tag as40bb47c8 --To-tag as1e663664 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 12537fc76492deb85aa7cd6072c3b4fb@159.65.48.104:5060 [Aug 18 10:34:29] DEBUG[15167] stasis.c: Creating topic. name: cache:693/bridge:8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95, detail: [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[13947] stasis/control.c: robot_213007, beb17a84-adfc-4fa3-b7a8-31977a540c1f: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13947] stasis/app.c: bridge 'beb17a84-adfc-4fa3-b7a8-31977a540c1f': is 2 interested in calls_0 [Aug 18 10:34:29] DEBUG[13947] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:29] DEBUG[13905] stasis/control.c: robot_213007: Channel departing bridge [Aug 18 10:34:29] DEBUG[13905] bridge.c: Waiting for 0x7f0c180c42f0(UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00) bridge thread to die. [Aug 18 10:34:29] DEBUG[13905] stasis/app.c: channel 'robot_213007': is 1 interested in calls_0 [Aug 18 10:34:29] DEBUG[13905] channel.c: Channel 0x7f0c34028b90 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00' hanging up. Refs: 2 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #155 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #155)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #157 (2) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #157)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (5) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909761 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:29] DEBUG[15169] bridge_channel.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: 0x7f0c8004d4e0(SIP/zvonobot-0000003f) is joining [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[15167] stasis.c: Topic 'cache:693/bridge:8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95': 0x7f0ca803b790 created [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #107 (6) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #107)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116886@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872 Max-Forwards: 70 From: ;tag=as7d998899 To: Contact: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 183885595 183885595 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14616 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (3) INVITE - 5 [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:29] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:29] DEBUG[14819] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:29] DEBUG[14819] http.c: HTTP closing session. Top level [Aug 18 10:34:29] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8;received=159.65.48.104 From: ;tag=as2f5156ef To: ;tag=as4a4114fd Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dd12f75" Content-Length: 0 <-------------> [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:29] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK551bd6c8;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2f5156ef [Aug 18 10:34:30] DEBUG[14671] channel.c: Channel 0x7f0c940f1490 'Snoop/212974-00000019' allocated [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4a4114fd [Aug 18 10:34:30] DEBUG[14392] bridge_native_rtp.c: Bridge '21515bb0-91f2-4ad5-852f-8721c870cad7' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14392] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[14392] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14392] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14392] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[14392] bridge.c: Bridge 21515bb0-91f2-4ad5-852f-8721c870cad7 is already using the new technology. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dd12f75" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2485aced650f4f671041baca16773141@159.65.48.104:5060 (Checking To) --From tag as2f5156ef --To-tag as4a4114fd [Aug 18 10:34:30] DEBUG[14671] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2485aced650f4f671041baca16773141@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #59 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[15167] bridge_native_rtp.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #59)) [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15167] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[15167] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:30] DEBUG[15167] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4;received=159.65.48.104 From: ;tag=as58ae887d To: ;tag=as62f9cfe9 Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c2e56a5" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as58ae887d [Aug 18 10:34:30] DEBUG[15167] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as62f9cfe9 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14814] channel.c: Channel 0x7f0c0807aab0 'Recorder/ARI-0000004c;2' allocated [Aug 18 10:34:30] DEBUG[14999] channel.c: Channel 0x7f0c20038420 'UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0' allocated [Aug 18 10:34:30] DEBUG[15169] bridge_channel.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: pushing 0x7f0c8004d4e0(SIP/zvonobot-0000003f) [Aug 18 10:34:30] DEBUG[15167] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: calling simple_bridge technology constructor [Aug 18 10:34:30] DEBUG[14963] app.c: One waitfor failed, trying another [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c2e56a5" [Aug 18 10:34:30] DEBUG[14999] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 (Checking To) --From tag as58ae887d --To-tag as62f9cfe9 [Aug 18 10:34:30] DEBUG[14814] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d8438f4 Max-Forwards: 70 From: ;tag=as58ae887d To: ;tag=as62f9cfe9 Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 16950 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #150 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[14999] res_rtp_asterisk.c: 0x7f0c200e7120 -- Strict RTP learning after remote address set to: 127.0.0.1:50282 [Aug 18 10:34:30] DEBUG[15174] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c0805d570(Recorder/ARI-0000004c;2) is joining [Aug 18 10:34:30] DEBUG[15171] stasis/app.c: Channel '1629282867.556' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15171] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 146 instead [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #78 (5) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #78)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398572 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[15169] bridge_channel.c: Channel SIP/zvonobot-0000003f joined 'simple_bridge' stasis-bridge [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #27)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[14221] threadpool.c: Worker thread idle timeout reached. Dying. [Aug 18 10:34:30] DEBUG[20523] threadpool.c: Destroying worker thread 12 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:30] DEBUG[15174] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: pushing 0x7f0c0805d570(Recorder/ARI-0000004c;2) [Aug 18 10:34:30] DEBUG[15167] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: calling simple_bridge technology start [Aug 18 10:34:30] DEBUG[15175] http.c: HTTP opening session. Top level [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7;received=159.65.48.104 From: ;tag=as1cccf2a3 To: ;tag=as09a6a224 Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a943c2f" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1cccf2a3 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as09a6a224 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a943c2f" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 (Checking To) --From tag as1cccf2a3 --To-tag as09a6a224 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #69 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47f2feb7 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: ;tag=as09a6a224 Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[14999] res_stasis.c: calls_0: Subscribing to robot_213006 [Aug 18 10:34:30] DEBUG[14999] stasis/app.c: Channel 'robot_213006' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15175] http.c: HTTP Request URI is /ari/channels/213206?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116834&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15169] bridge_native_rtp.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[15169] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[15169] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15179] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15169] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 16374 [Aug 18 10:34:30] DEBUG[15174] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[15174] bridge_channel.c: Channel Recorder/ARI-0000004c;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[15169] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15169] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34 is already using the new technology. [Aug 18 10:34:30] DEBUG[15179] http.c: HTTP Request URI is /ari/channels/213205?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116835&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15179] http.c: match request [ari/channels/213205] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15179] http.c: match request [ari/channels/213205] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15179] http.c: match request [ari/channels/213205] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[14999] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[14999] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] VERBOSE[15180] dial.c: Called 127.0.0.1:50282 [Aug 18 10:34:30] DEBUG[15179] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #159 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15179] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Finding handler for channels/213205 [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[15174] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad'. Checking compatability for channels 'SIP/zvonobot-00000034' and 'Recorder/ARI-0000004c;2' [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Finding handler for 213205 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking channels create: Didn't match 213205 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15179] res_ari.c: Checking channels externalMedia: Didn't match 213205 [Aug 18 10:34:30] DEBUG[15179] res_ari.c: No explicit handler found for 213205. Using wildcard channelId. [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6407ms with no response [Aug 18 10:34:30] DEBUG[15174] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' can not use native RTP bridge as could not get details [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Hanging up call 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:30] DEBUG[15174] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[15174] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15174] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15174] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15174] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad is already using the new technology. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:30] DEBUG[15174] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c0805d570(Recorder/ARI-0000004c;2) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15184] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15184] http.c: HTTP Request URI is /ari/channels/213204?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116836&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15184] http.c: match request [ari/channels/213204] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15184] http.c: match request [ari/channels/213204] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15184] http.c: match request [ari/channels/213204] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15184] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15184] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Finding handler for channels/213204 [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Finding handler for 213204 [Aug 18 10:34:30] DEBUG[14813] channel.c: Channel 0x7f0c100541e0 'Snoop/213015-0000001a' allocated [Aug 18 10:34:30] DEBUG[15167] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[14112] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83'. Checking compatability for channels 'SIP/zvonobot-00000033' and 'Recorder/ARI-0000002d;2' [Aug 18 10:34:30] DEBUG[14112] bridge_native_rtp.c: Bridge '3fc9ee09-2746-49ab-833c-6c9b37b1bb83' can not use native RTP bridge as channel 'SIP/zvonobot-00000033' has features which prevent it [Aug 18 10:34:30] DEBUG[14112] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[14112] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14112] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14112] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[14112] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 is already using the new technology. [Aug 18 10:34:30] DEBUG[15169] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: 0x7f0c8004d4e0(SIP/zvonobot-0000003f) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15175] http.c: match request [ari/channels/213206] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15174] channel.c: Channel Recorder/ARI-0000004c;2 setting read format path: slin -> slin [Aug 18 10:34:30] DEBUG[15187] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15174] channel.c: Channel SIP/zvonobot-00000034 setting write format path: slin -> alaw [Aug 18 10:34:30] DEBUG[15185] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15187] http.c: HTTP Request URI is /ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/addChannel?channel=213043 [Aug 18 10:34:30] DEBUG[15187] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15187] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15187] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/addChannel] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15187] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Finding handler for bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/addChannel [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15185] http.c: HTTP Request URI is /ari/channels/1629282830.62 [Aug 18 10:34:30] DEBUG[14830] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:30] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000be - start 1629282863.614540 answer 0.000000 end 1629282870.100502 dur 6.485 bill 1629282870.100 dispo NO ANSWER [Aug 18 10:34:30] DEBUG[14830] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15167] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15174] channel.c: Channel SIP/zvonobot-00000034 setting read format path: alaw -> slin [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c;received=159.65.48.104 From: ;tag=as34e313e7 To: ;tag=as3cc5431f Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 301417948 301417948 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15990 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6fb4472c;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as34e313e7 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3cc5431f [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 301417948 301417948 IN IP4 178.62.121.41 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:30] VERBOSE[15180] dial.c: UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 answered [Aug 18 10:34:30] VERBOSE[15180] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 [Aug 18 10:34:30] DEBUG[15185] http.c: match request [ari/channels/1629282830.62] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[14813] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking channels create: Didn't match 213204 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15990 RTP/AVP 0 8 101 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:30] DEBUG[15174] channel.c: Channel Recorder/ARI-0000004c;2 setting write format path: slin -> slin [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:30] DEBUG[14827] channel.c: Channel 0x7f0c18029cf0 'SIP/zvonobot-000000be' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[15185] http.c: match request [ari/channels/1629282830.62] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:30] DEBUG[15193] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[14813] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15185] http.c: match request [ari/channels/1629282830.62] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15175] http.c: match request [ari/channels/213206] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Finding handler for 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:30] DEBUG[15185] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:30] DEBUG[15186] stasis/app.c: Channel '1629282867.561' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15186] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 145 instead [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15187] res_ari.c: No explicit handler found for 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Finding handler for addChannel [Aug 18 10:34:30] DEBUG[15187] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:30] DEBUG[15187] stasis/control.c: 213043: Sending channel add_to_bridge command [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 (Checking To) --From tag as34e313e7 --To-tag as3cc5431f [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 59976a1b2b58a03740e33f604daad9f1@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 825 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188;received=159.65.48.104 From: ;tag=as45eb6124 To: ;tag=as22bf2ab4 Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e902bc1" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as45eb6124 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as22bf2ab4 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e902bc1" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 (Checking To) --From tag as45eb6124 --To-tag as22bf2ab4 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #67 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78c71188 Max-Forwards: 70 From: ;tag=as45eb6124 To: ;tag=as22bf2ab4 Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Finding handler for channels/1629282830.62 [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Finding handler for 1629282830.62 [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking channels create: Didn't match 1629282830.62 [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15185] res_ari.c: Checking channels externalMedia: Didn't match 1629282830.62 [Aug 18 10:34:30] DEBUG[15185] res_ari.c: No explicit handler found for 1629282830.62. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15193] http.c: HTTP Request URI is /ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/play?media=sound%3Asilence%2F2 [Aug 18 10:34:30] DEBUG[15169] res_rtp_asterisk.c: (0x7f0c30074490) RTP changing ssrc from 1449108826 to 1803408674 due to a source change [Aug 18 10:34:30] DEBUG[15161] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:30] DEBUG[15161] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[13416] stasis/app.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34' is 2 interested in calls_0 [Aug 18 10:34:30] DEBUG[15195] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15175] http.c: match request [ari/channels/213206] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[15184] res_ari.c: Checking channels externalMedia: Didn't match 213204 [Aug 18 10:34:30] DEBUG[15195] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213015&app=calls_0&format=slin16&external_host=127.0.0.1%3A50315 [Aug 18 10:34:30] DEBUG[15184] res_ari.c: No explicit handler found for 213204. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15197] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15198] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15175] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 11726 [Aug 18 10:34:30] DEBUG[15197] http.c: HTTP Request URI is /ari/channels/213208?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116832&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15193] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/play] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15193] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/play] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15193] http.c: match request [ari/bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/play] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15193] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[13568] bridge_roles.c: Roles did not exist on channel SIP/zvonobot-0000004d [Aug 18 10:34:30] DEBUG[13568] stasis/control.c: 213043: Adding to bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[15198] http.c: HTTP Request URI is /ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/record?name=213032_vEJKgxiyRBaoxkURsNCEteyerFelGHYd&format=wav [Aug 18 10:34:30] DEBUG[13568] stasis/app.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15195] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Finding handler for bridges/3fc9ee09-2746-49ab-833c-6c9b37b1bb83/play [Aug 18 10:34:30] DEBUG[15199] http.c: HTTP opening session. Top level [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #75 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #146 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #146)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #150 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #150)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 From: ;tag=as671c682b To: ;tag=as53b926c5 Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK36a1094b;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as671c682b [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as53b926c5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="290463d2" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 (Checking To) --From tag as671c682b --To-tag as53b926c5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2749fa7d41ec862f1556002a63546011@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] DEBUG[14112] audiohook.c: Audiohook 0x7f0c1014c260 has stale audio in its factories. Flushing them both [Aug 18 10:34:30] DEBUG[15175] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15197] http.c: match request [ari/channels/213208] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Finding handler for channels/213206 [Aug 18 10:34:30] DEBUG[15198] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/record] with handler [httpstatus] len 10 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2;received=159.65.48.104 From: ;tag=as1cc5f222 To: ;tag=as46a80351 Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259c04d7" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK397461c2;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1cc5f222 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as46a80351 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259c04d7" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 (Checking To) --From tag as1cc5f222 --To-tag as46a80351 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 517fac1c16b061d94e7352d96f8e8dbf@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 From: ;tag=as1c2a52a2 To: ;tag=as025c937f Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1c2a52a2 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as025c937f [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 (Checking To) --From tag as1c2a52a2 --To-tag as025c937f [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116888@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe Max-Forwards: 70 From: ;tag=as1c2a52a2 To: Contact: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9;received=159.65.48.104 From: ;tag=as69c6d00c To: ;tag=as56e7ef31 Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67379e21" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as69c6d00c [Aug 18 10:34:30] DEBUG[15198] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/record] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15195] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15199] http.c: HTTP Request URI is /ari/channels/213207?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116833&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56e7ef31 [Aug 18 10:34:30] DEBUG[15197] http.c: match request [ari/channels/213208] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15199] http.c: match request [ari/channels/213207] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15198] http.c: match request [ari/bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/record] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[14814] res_stasis_recording.c: 1629282863.500: Sending record(213016_MtjOSnOVSDieYyOzUQyEuUjYfOSqPhfe.wav) command [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15195] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15195] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15200] bridge_channel.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: 0x7f0c940389d0(SIP/zvonobot-0000004d) is joining [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15197] http.c: match request [ari/channels/213208] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15199] http.c: match request [ari/channels/213207] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15198] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15199] http.c: match request [ari/channels/213207] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:30] DEBUG[15180] stasis/app.c: Channel 'robot_213006' is 2 interested in calls_0 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15197] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15199] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Finding handler for 213206 [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Finding handler for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking channels create: Didn't match 213206 [Aug 18 10:34:30] DEBUG[15199] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15197] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15202] app.c: play_and_record: , /var/spool/asterisk/recording/213016_MtjOSnOVSDieYyOzUQyEuUjYfOSqPhfe, 'wav' [Aug 18 10:34:30] DEBUG[15202] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:30] VERBOSE[15202] app.c: x=0, open writing: /var/spool/asterisk/recording/213016_MtjOSnOVSDieYyOzUQyEuUjYfOSqPhfe format: wav, 0x7f0c3c007240 [Aug 18 10:34:30] DEBUG[15203] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15203] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[15203] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[14654] channel.c: Channel 0x7f0c78069a80 'Recorder/ARI-00000034;2' destroying [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[14814] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: Checking channels externalMedia: Didn't match 213206 [Aug 18 10:34:30] DEBUG[14998] channel.c: Channel 0x7f0c241243f0 'UnicastRTP/127.0.0.1:50312-0x7f0c240eada0' allocated [Aug 18 10:34:30] DEBUG[14814] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Finding handler for channels/213207 [Aug 18 10:34:30] DEBUG[15175] res_ari.c: No explicit handler found for 213206. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15203] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15193] res_ari.c: No explicit handler found for 3fc9ee09-2746-49ab-833c-6c9b37b1bb83. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[14998] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:30] VERBOSE[14998] res_rtp_asterisk.c: 0x7f0c24067830 -- Strict RTP learning after remote address set to: 127.0.0.1:50312 [Aug 18 10:34:30] DEBUG[14998] res_stasis.c: calls_0: Subscribing to robot_212996 [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15003] channel.c: Channel 0x7f0c301165c0 'Recorder/ARI-00000050;1' allocated [Aug 18 10:34:30] DEBUG[15003] stasis.c: Creating topic. name: channel:1629282870.604, detail: [Aug 18 10:34:30] DEBUG[15003] stasis.c: Topic 'channel:1629282870.604': 0x7f0c300abff0 created [Aug 18 10:34:30] DEBUG[15003] stasis.c: Creating topic. name: cache:694/channel:1629282870.604, detail: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67379e21" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[14998] stasis/app.c: Channel 'robot_212996' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 (Checking To) --From tag as69c6d00c --To-tag as56e7ef31 [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] http.c: HTTP Request URI is /ari/channels/213209?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116831&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Finding handler for play [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:30] DEBUG[15193] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:30] DEBUG[15193] stasis.c: Creating topic. name: channel:1629282870.605, detail: [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15193] stasis.c: Topic 'channel:1629282870.605': 0x7f0c180d3690 created [Aug 18 10:34:30] DEBUG[15193] stasis.c: Creating topic. name: cache:695/channel:1629282870.605, detail: [Aug 18 10:34:30] DEBUG[15203] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Finding handler for bridges/ef2f0ed6-7b56-46c3-a894-dc0114b2fb34/record [Aug 18 10:34:30] DEBUG[15205] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15003] stasis.c: Topic 'cache:694/channel:1629282870.604': 0x7f0c300414a0 created [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15193] stasis.c: Topic 'cache:695/channel:1629282870.605': 0x7f0c1800a330 created [Aug 18 10:34:30] DEBUG[15201] http.c: match request [ari/channels/213209] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15200] bridge_channel.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: pushing 0x7f0c940389d0(SIP/zvonobot-0000004d) [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[14998] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Finding handler for channels/213208 [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] VERBOSE[15200] bridge_channel.c: Channel SIP/zvonobot-0000004d joined 'simple_bridge' stasis-bridge <8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95> [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[14998] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15203] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6df71ee9 Max-Forwards: 70 From: ;tag=as69c6d00c To: ;tag=as56e7ef31 Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 16210 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Finding handler for 213207 [Aug 18 10:34:30] DEBUG[15204] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15205] http.c: HTTP Request URI is /ari/channels/213210?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116830&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking channels create: Didn't match 213207 [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15201] http.c: match request [ari/channels/213209] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15204] http.c: HTTP Request URI is /ari/bridges/9044ffbe-38f0-4f08-9b7d-173767b0d858/addChannel?channel=1629282863.496%2Crobot_213006 [Aug 18 10:34:30] DEBUG[15200] bridge_native_rtp.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[15200] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[15200] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15200] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15200] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15200] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95 is already using the new technology. [Aug 18 10:34:30] DEBUG[15200] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: 0x7f0c940389d0(SIP/zvonobot-0000004d) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15200] res_rtp_asterisk.c: (0x2c36530) RTP changing ssrc from 1766902072 to 946450846 due to a source change [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[13568] stasis/app.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95' is 2 interested in calls_0 [Aug 18 10:34:30] DEBUG[15207] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15205] http.c: match request [ari/channels/213210] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15201] http.c: match request [ari/channels/213209] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15187] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:30] DEBUG[15207] http.c: HTTP Request URI is /ari/channels/213211?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116829&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15207] http.c: match request [ari/channels/213211] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15207] http.c: match request [ari/channels/213211] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15207] http.c: match request [ari/channels/213211] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15207] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15207] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Finding handler for channels/213211 [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Finding handler for 213211 [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking channels create: Didn't match 213211 [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15207] res_ari.c: Checking channels externalMedia: Didn't match 213211 [Aug 18 10:34:30] DEBUG[15207] res_ari.c: No explicit handler found for 213211. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15203] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15203] stasis.c: Creating topic. name: bridge:8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e, detail: [Aug 18 10:34:30] DEBUG[15203] stasis.c: Topic 'bridge:8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e': 0x7f0c38036d20 created [Aug 18 10:34:30] DEBUG[15203] stasis.c: Creating topic. name: cache:696/bridge:8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e, detail: [Aug 18 10:34:30] DEBUG[15203] stasis.c: Topic 'cache:696/bridge:8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e': 0x7f0c38006cd0 created [Aug 18 10:34:30] DEBUG[15203] bridge_native_rtp.c: Bridge '8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[15201] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15187] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15195] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:30] DEBUG[15195] netsock2.c: Splitting '127.0.0.1:50315' into... [Aug 18 10:34:30] DEBUG[15195] netsock2.c: ...host '127.0.0.1' and port '50315'. [Aug 18 10:34:30] DEBUG[15195] netsock2.c: Splitting '127.0.0.1:50315' into... [Aug 18 10:34:30] DEBUG[15195] netsock2.c: ...host '127.0.0.1' and port '50315'. [Aug 18 10:34:30] DEBUG[15195] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:30] DEBUG[15195] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c240f1de0' [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) RTP allocated port 14382 [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) ICE creating session 127.0.0.1:14382 (14382) [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) ICE create [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) ICE add system candidates [Aug 18 10:34:30] DEBUG[15195] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15195] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) ICE add candidate: 159.65.48.104:14382, 2130706431 [Aug 18 10:34:30] DEBUG[15195] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15195] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15195] res_rtp_asterisk.c: (0x7f0c240f1de0) ICE add candidate: 10.131.0.10:14382, 2130706431 [Aug 18 10:34:30] DEBUG[15195] rtp_engine.c: RTP instance '0x7f0c240f1de0' is setup and ready to go [Aug 18 10:34:30] DEBUG[15195] stasis.c: Creating topic. name: channel:robot_213015, detail: [Aug 18 10:34:30] DEBUG[15195] stasis.c: Topic 'channel:robot_213015': 0x7f0c24006d10 created [Aug 18 10:34:30] DEBUG[15195] stasis.c: Creating topic. name: cache:697/channel:robot_213015, detail: [Aug 18 10:34:30] DEBUG[15195] stasis.c: Topic 'cache:697/channel:robot_213015': 0x7f0c24051fe0 created [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Finding handler for channels/213209 [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Finding handler for 213209 [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking channels create: Didn't match 213209 [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15201] res_ari.c: Checking channels externalMedia: Didn't match 213209 [Aug 18 10:34:30] DEBUG[15201] res_ari.c: No explicit handler found for 213209. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15199] res_ari.c: Checking channels externalMedia: Didn't match 213207 [Aug 18 10:34:30] DEBUG[15199] res_ari.c: No explicit handler found for 213207. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15205] http.c: match request [ari/channels/213210] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #159 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15210] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Finding handler for 213208 [Aug 18 10:34:30] VERBOSE[15206] dial.c: Called 127.0.0.1:50312 [Aug 18 10:34:30] DEBUG[15210] http.c: HTTP Request URI is /ari/channels/213212?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116828&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: Allocating new SIP dialog for 773d0d5662241c155243874a601f5704@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15184] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2c9f0c0' [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) RTP allocated port 12210 [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE creating session 0.0.0.0:12210 (12210) [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE create [Aug 18 10:34:30] DEBUG[15208] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15210] http.c: match request [ari/channels/213212] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[14654] stasis.c: Destroying topic. name: cache:444/channel:1629282856.386, detail: [Aug 18 10:34:30] DEBUG[14654] stasis.c: Topic 'cache:444/channel:1629282856.386': 0x7f0c7803d9e0 destroyed [Aug 18 10:34:30] DEBUG[14654] stasis.c: Destroying topic. name: channel:1629282856.386, detail: [Aug 18 10:34:30] DEBUG[14654] stasis.c: Topic 'channel:1629282856.386': 0x7f0c78070280 destroyed [Aug 18 10:34:30] VERBOSE[15206] dial.c: UnicastRTP/127.0.0.1:50312-0x7f0c240eada0 answered [Aug 18 10:34:30] DEBUG[15205] http.c: match request [ari/channels/213210] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE add system candidates [Aug 18 10:34:30] DEBUG[15184] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15184] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE add candidate: 159.65.48.104:12210, 2130706431 [Aug 18 10:34:30] DEBUG[15184] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15184] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE add candidate: 10.131.0.10:12210, 2130706431 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:30] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking channels create: Didn't match 213208 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:30] DEBUG[15210] http.c: match request [ari/channels/213212] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15184] rtp_engine.c: RTP instance '0x2c9f0c0' is setup and ready to go [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #159)) [Aug 18 10:34:30] DEBUG[15204] http.c: match request [ari/bridges/9044ffbe-38f0-4f08-9b7d-173767b0d858/addChannel] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15211] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) ICE stopped [Aug 18 10:34:30] DEBUG[15184] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15184] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15184] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15184] res_rtp_asterisk.c: (0x2c9f0c0) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15184] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15184] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15184] chan_sip.c: SIP call-id changed from '773d0d5662241c155243874a601f5704@127.0.1.1:5060' to '64d2aac75896f6d3140509d05f605000@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15184] stasis.c: Creating topic. name: channel:213204, detail: [Aug 18 10:34:30] DEBUG[15184] stasis.c: Topic 'channel:213204': 0x2c6c2b0 created [Aug 18 10:34:30] DEBUG[15184] stasis.c: Creating topic. name: cache:698/channel:213204, detail: [Aug 18 10:34:30] DEBUG[15184] stasis.c: Topic 'cache:698/channel:213204': 0x2c590f0 created [Aug 18 10:34:30] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] VERBOSE[15206] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50312-0x7f0c240eada0 [Aug 18 10:34:30] DEBUG[15206] stasis/app.c: Channel 'robot_212996' is 2 interested in calls_0 [Aug 18 10:34:30] DEBUG[14720] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:30] DEBUG[15018] channel.c: Channel 0x7f0c9c095d60 'Recorder/ARI-00000051;1' allocated [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15018] stasis.c: Creating topic. name: channel:1629282870.608, detail: [Aug 18 10:34:30] DEBUG[15018] stasis.c: Topic 'channel:1629282870.608': 0x7f0c9c025420 created [Aug 18 10:34:30] DEBUG[15018] stasis.c: Creating topic. name: cache:699/channel:1629282870.608, detail: [Aug 18 10:34:30] DEBUG[15018] stasis.c: Topic 'cache:699/channel:1629282870.608': 0x7f0c9c096a20 created [Aug 18 10:34:30] DEBUG[13329] channel.c: Channel 0x7f0c88047e20 'Recorder/ARI-0000000a;2' destroying [Aug 18 10:34:30] DEBUG[12962] chan_sip.c: Hangup call SIP/zvonobot-00000010, SIP callid 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[12962] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[12962] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14853] channel.c: Channel 0x7f0c9803e700 'Recorder/ARI-0000004d;2' allocated [Aug 18 10:34:30] DEBUG[14853] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:30] DEBUG[14545] chan_sip.c: Hangup call SIP/zvonobot-000000a4, SIP callid 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15204] http.c: match request [ari/bridges/9044ffbe-38f0-4f08-9b7d-173767b0d858/addChannel] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15205] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15205] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Finding handler for channels/213210 [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15203] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[14739] chan_sip.c: Hangup call SIP/zvonobot-000000b5, SIP callid 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14739] res_rtp_asterisk.c: (0x7f0cb4085d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[13340] channel.c: Channel 0x7f0ca0046fc0 'Announcer/ARI-0000000b;2' destroying [Aug 18 10:34:30] DEBUG[14720] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[14736] chan_sip.c: Hangup call SIP/zvonobot-000000b4, SIP callid 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15208] http.c: HTTP Request URI is /ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/record?name=213043_egRKVPBnNeSLLBzTdbfjfjBSRIkzUZEb&format=wav [Aug 18 10:34:30] DEBUG[14739] res_rtp_asterisk.c: (0x7f0cb4085d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15204] http.c: match request [ari/bridges/9044ffbe-38f0-4f08-9b7d-173767b0d858/addChannel] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[14545] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:30] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] VERBOSE[13330] app.c: User hung up [Aug 18 10:34:30] DEBUG[13330] res_stasis_recording.c: 1629282833.91: Recording complete [Aug 18 10:34:30] DEBUG[13330] channel.c: Channel 0x7f0c88037560 'Recorder/ARI-0000000a;1' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[14739] channel.c: Channel 0x7f0cb4024d90 'SIP/zvonobot-000000b5' destroying [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[14736] res_rtp_asterisk.c: (0x7f0c1c14b660) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14736] res_rtp_asterisk.c: (0x7f0c1c14b660) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14736] channel.c: Channel 0x7f0c1c09dfb0 'SIP/zvonobot-000000b4' destroying [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[14545] res_rtp_asterisk.c: (0x7f0cb0161a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[13329] stasis.c: Destroying topic. name: cache:110/channel:1629282833.92, detail: [Aug 18 10:34:30] DEBUG[15211] http.c: HTTP Request URI is /ari/channels/213213?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116827&callerId=74950493843 [Aug 18 10:34:30] DEBUG[15210] http.c: match request [ari/channels/213212] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15208] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/record] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[13351] bridge_channel.c: Setting 0x7f0c9804a330(Snoop/212981-00000004) state from:0 to:1 [Aug 18 10:34:30] DEBUG[13351] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: pulling 0x7f0c9804a330(Snoop/212981-00000004) [Aug 18 10:34:30] VERBOSE[13351] bridge_channel.c: Channel Snoop/212981-00000004 left 'simple_bridge' stasis-bridge <25e1770d-58e8-4da7-94aa-19844c10fa1c> [Aug 18 10:34:30] DEBUG[13351] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0c9804a330(Snoop/212981-00000004) is leaving simple_bridge technology [Aug 18 10:34:30] DEBUG[13351] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[13351] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[13351] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[13351] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[13351] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[13351] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c is already using the new technology. [Aug 18 10:34:30] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15197] res_ari.c: Checking channels externalMedia: Didn't match 213208 [Aug 18 10:34:30] DEBUG[15197] res_ari.c: No explicit handler found for 213208. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15212] bridge_channel.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: 0x7f0c980ad0d0(Recorder/ARI-0000004d;2) is joining [Aug 18 10:34:30] DEBUG[15204] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[13340] stasis.c: Destroying topic. name: cache:114/channel:1629282833.95, detail: [Aug 18 10:34:30] DEBUG[13340] stasis.c: Topic 'cache:114/channel:1629282833.95': 0x7f0ca0046cd0 destroyed [Aug 18 10:34:30] DEBUG[13340] stasis.c: Destroying topic. name: channel:1629282833.95, detail: [Aug 18 10:34:30] DEBUG[13340] stasis.c: Topic 'channel:1629282833.95': 0x7f0ca0048d00 destroyed [Aug 18 10:34:30] DEBUG[13329] stasis.c: Topic 'cache:110/channel:1629282833.92': 0x7f0c88049ce0 destroyed [Aug 18 10:34:30] DEBUG[13329] stasis.c: Destroying topic. name: channel:1629282833.92, detail: [Aug 18 10:34:30] DEBUG[13329] stasis.c: Topic 'channel:1629282833.92': 0x7f0c88049ad0 destroyed [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[14545] res_rtp_asterisk.c: (0x7f0cb0161a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14545] channel.c: Channel 0x7f0cb011fc00 'SIP/zvonobot-000000a4' destroying [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #59 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #59)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #142 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #142)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15203] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15210] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[15208] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/record] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:30] DEBUG[13351] bridge_channel.c: Bridge is returning 0x7f0c9804a330(Snoop/212981-00000004) to read format slin [Aug 18 10:34:30] DEBUG[13351] channel.c: Channel Snoop/212981-00000004 setting read format path: slin -> slin [Aug 18 10:34:30] DEBUG[13351] bridge_channel.c: Bridge is returning 0x7f0c9804a330(Snoop/212981-00000004) to write format slin [Aug 18 10:34:30] DEBUG[13351] channel.c: Channel Snoop/212981-00000004 setting write format path: slin -> slin [Aug 18 10:34:30] DEBUG[13351] stasis/control.c: 1629282833.93, 25e1770d-58e8-4da7-94aa-19844c10fa1c: Channel was departed from bridge [Aug 18 10:34:30] DEBUG[13351] stasis/app.c: bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c': is 3 interested in calls_0 [Aug 18 10:34:30] DEBUG[13351] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:30] DEBUG[13333] stasis/control.c: 1629282833.93: Channel departing bridge [Aug 18 10:34:30] DEBUG[13333] bridge.c: Waiting for 0x7f0c9804a330(Snoop/212981-00000004) bridge thread to die. [Aug 18 10:34:30] DEBUG[13333] stasis/app.c: channel '1629282833.93': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[13333] stasis/app.c: channel '1629282833.93' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[13333] channel.c: Channel 0x7f0c9c0305a0 'Snoop/212981-00000004' hanging up. Refs: 3 [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: Allocating new SIP dialog for 0d175bd467b9fd7308d9673e61239679@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15175] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0047010' [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) RTP allocated port 10426 [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[14244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE creating session 0.0.0.0:10426 (10426) [Aug 18 10:34:30] DEBUG[15208] http.c: match request [ari/bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/record] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE create [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE add system candidates [Aug 18 10:34:30] DEBUG[15175] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15175] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE add candidate: 159.65.48.104:10426, 2130706431 [Aug 18 10:34:30] DEBUG[15175] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15175] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE add candidate: 10.131.0.10:10426, 2130706431 [Aug 18 10:34:30] DEBUG[15175] rtp_engine.c: RTP instance '0x7f0cb0047010' is setup and ready to go [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) ICE stopped [Aug 18 10:34:30] DEBUG[15175] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15175] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15208] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15175] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15175] res_rtp_asterisk.c: (0x7f0cb0047010) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15175] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15175] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15175] chan_sip.c: SIP call-id changed from '0d175bd467b9fd7308d9673e61239679@127.0.1.1:5060' to '71babb7e3832ba1353344961206841b0@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15175] stasis.c: Creating topic. name: channel:213206, detail: [Aug 18 10:34:30] DEBUG[15175] stasis.c: Topic 'channel:213206': 0x7f0cb01b8c80 created [Aug 18 10:34:30] DEBUG[15175] stasis.c: Creating topic. name: cache:700/channel:213206, detail: [Aug 18 10:34:30] DEBUG[15175] stasis.c: Topic 'cache:700/channel:213206': 0x7f0cb00df550 created [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Finding handler for bridges/8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95/record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Finding handler for 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95 [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15208] res_ari.c: No explicit handler found for 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Finding handler for record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:30] DEBUG[15208] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:30] DEBUG[15208] stasis.c: Creating topic. name: channel:1629282870.611, detail: [Aug 18 10:34:30] DEBUG[15208] stasis.c: Topic 'channel:1629282870.611': 0x7f0c7803d9e0 created [Aug 18 10:34:30] DEBUG[15208] stasis.c: Creating topic. name: cache:701/channel:1629282870.611, detail: [Aug 18 10:34:30] DEBUG[15208] stasis.c: Topic 'cache:701/channel:1629282870.611': 0x7f0c7803dac0 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.609, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.609': 0x7f0c300a91a0 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:702/channel:1629282870.609, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:702/channel:1629282870.609': 0x7f0c3004d1c0 created [Aug 18 10:34:30] DEBUG[15210] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:702/channel:1629282870.609, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:702/channel:1629282870.609': 0x7f0c3004d1c0 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.609, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.609': 0x7f0c300a91a0 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:20', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b4', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213153', '')] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 From: ;tag=as7dd13c21 To: ;tag=as19c9362c Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7dd13c21 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as19c9362c [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 (Checking To) --From tag as7dd13c21 --To-tag as19c9362c [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15203] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:30] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212981-00000004 - start 1629282833.408416 answer 1629282833.408416 end 1629282870.317899 dur 36.909 bill 36.909 dispo ANSWERED [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.612, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.612': 0x7f0c3007f1a0 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:703/channel:1629282870.612, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:703/channel:1629282870.612': 0x7f0c300a9240 created [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Finding handler for bridges/9044ffbe-38f0-4f08-9b7d-173767b0d858/addChannel [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Finding handler for ef2f0ed6-7b56-46c3-a894-dc0114b2fb34 [Aug 18 10:34:30] DEBUG[15203] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15211] http.c: match request [ari/channels/213213] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15198] res_ari.c: No explicit handler found for ef2f0ed6-7b56-46c3-a894-dc0114b2fb34. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Finding handler for record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:30] DEBUG[15198] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:30] DEBUG[15198] stasis.c: Creating topic. name: channel:1629282870.613, detail: [Aug 18 10:34:30] DEBUG[15198] stasis.c: Topic 'channel:1629282870.613': 0x7f0c2c0abac0 created [Aug 18 10:34:30] DEBUG[15198] stasis.c: Creating topic. name: cache:704/channel:1629282870.613, detail: [Aug 18 10:34:30] DEBUG[15198] stasis.c: Topic 'cache:704/channel:1629282870.613': 0x7f0c2c018280 created [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Finding handler for 213210 [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking channels create: Didn't match 213210 [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15205] res_ari.c: Checking channels externalMedia: Didn't match 213210 [Aug 18 10:34:30] DEBUG[15205] res_ari.c: No explicit handler found for 213210. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Finding handler for channels/213212 [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: Allocating new SIP dialog for 05c5c30c69e0a30b68c302b13fbcb61a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15179] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac06c160' [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) RTP allocated port 17162 [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE creating session 0.0.0.0:17162 (17162) [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE create [Aug 18 10:34:30] DEBUG[15212] bridge_channel.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: pushing 0x7f0c980ad0d0(Recorder/ARI-0000004d;2) [Aug 18 10:34:30] DEBUG[15203] bridge.c: Bridge 8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e: calling simple_bridge technology constructor [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE add system candidates [Aug 18 10:34:30] DEBUG[15179] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15179] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE add candidate: 159.65.48.104:17162, 2130706431 [Aug 18 10:34:30] DEBUG[15179] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15179] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE add candidate: 10.131.0.10:17162, 2130706431 [Aug 18 10:34:30] DEBUG[15179] rtp_engine.c: RTP instance '0x7f0cac06c160' is setup and ready to go [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) ICE stopped [Aug 18 10:34:30] DEBUG[15179] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15179] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15179] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15211] http.c: match request [ari/channels/213213] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #150 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #150)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #153 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #153)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #155 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #155)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6368ms with no response [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Hanging up call 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14843] channel.c: Channel 0x7f0c400c7ad0 'SIP/zvonobot-000000c1' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #157 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #157)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15203] bridge.c: Bridge 8e36fc0e-5cfe-43fc-8c1f-77a350dbfb3e: calling simple_bridge technology start [Aug 18 10:34:30] DEBUG[15213] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15179] res_rtp_asterisk.c: (0x7f0cac06c160) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15179] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15179] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15211] http.c: match request [ari/channels/213213] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15179] chan_sip.c: SIP call-id changed from '05c5c30c69e0a30b68c302b13fbcb61a@127.0.1.1:5060' to '533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15179] stasis.c: Creating topic. name: channel:213205, detail: [Aug 18 10:34:30] DEBUG[15179] stasis.c: Topic 'channel:213205': 0x7f0cac043a80 created [Aug 18 10:34:30] DEBUG[15179] stasis.c: Creating topic. name: cache:705/channel:213205, detail: [Aug 18 10:34:30] DEBUG[15179] stasis.c: Topic 'cache:705/channel:213205': 0x7f0cac0443d0 created [Aug 18 10:34:30] DEBUG[15213] http.c: HTTP Request URI is /ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record?name=213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr&format=wav [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Finding handler for 9044ffbe-38f0-4f08-9b7d-173767b0d858 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[15213] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15211] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15204] res_ari.c: No explicit handler found for 9044ffbe-38f0-4f08-9b7d-173767b0d858. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:703/channel:1629282870.612, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:703/channel:1629282870.612': 0x7f0c300a9240 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.612, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.612': 0x7f0c3007f1a0 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:17', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a4', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213130', '')] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Finding handler for addChannel [Aug 18 10:34:30] DEBUG[15213] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.615, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.615': 0x7f0c300e3b20 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:706/channel:1629282870.615, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:706/channel:1629282870.615': 0x7f0c300abee0 created [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: Allocating new SIP dialog for 75d72a2f4425b0714f3f80a318268fea@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15199] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c280a3c30' [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) RTP allocated port 14362 [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE creating session 0.0.0.0:14362 (14362) [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE create [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15204] res_ari.c: Checking bridgeId addChannel: Explicit match with addChannel [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213153': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Finding handler for 213212 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '76c394087cbc0dd73ea07fa80dca99ab@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4085d30) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4085d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4085d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb4085d30) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb4085d30' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '0dd52c9956dc6b544156ac634129900b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c14b660) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c14b660) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c14b660) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1c14b660) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c1c14b660' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '233e7fbe2f9b48a711e029b361c67849@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0161a90) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0161a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0161a90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0161a90) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb0161a90' [Aug 18 10:34:30] DEBUG[15213] http.c: match request [ari/bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record] with handler [ari] len 3 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 From: ;tag=as6f697fd0 To: ;tag=as48575d5d Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 20]: SIP/2.0 603 Declined [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6f697fd0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as48575d5d [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 (Checking To) --From tag as6f697fd0 --To-tag as48575d5d [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117003@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01318e68 Max-Forwards: 70 From: ;tag=as6f697fd0 To: ;tag=as48575d5d Contact: Call-ID: 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #159 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #159)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:30] DEBUG[15204] stasis/control.c: 1629282863.496: Sending channel add_to_bridge command [Aug 18 10:34:30] DEBUG[15203] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[14988] bridge_roles.c: Roles did not exist on channel Snoop/213006-00000016 [Aug 18 10:34:30] DEBUG[14988] stasis/control.c: 1629282863.496: Adding to bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213153' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:706/channel:1629282870.615, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:706/channel:1629282870.615': 0x7f0c300abee0 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.615, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.615': 0x7f0c300e3b20 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b5', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213150', '')] [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE add system candidates [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking channels create: Didn't match 213212 [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: Allocating new SIP dialog for 3fba10e261fa8b1121096ac47a08da7f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15207] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c00f100' [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) RTP allocated port 13902 [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE creating session 0.0.0.0:13902 (13902) [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE create [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE add system candidates [Aug 18 10:34:30] DEBUG[15207] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15207] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE add candidate: 159.65.48.104:13902, 2130706431 [Aug 18 10:34:30] DEBUG[15207] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15207] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE add candidate: 10.131.0.10:13902, 2130706431 [Aug 18 10:34:30] DEBUG[15207] rtp_engine.c: RTP instance '0x7f0c7c00f100' is setup and ready to go [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) ICE stopped [Aug 18 10:34:30] DEBUG[15207] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15207] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15207] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15207] res_rtp_asterisk.c: (0x7f0c7c00f100) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15207] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15207] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15207] chan_sip.c: SIP call-id changed from '3fba10e261fa8b1121096ac47a08da7f@127.0.1.1:5060' to '1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15207] stasis.c: Creating topic. name: channel:213211, detail: [Aug 18 10:34:30] DEBUG[15207] stasis.c: Topic 'channel:213211': 0x7f0c7c0bbd10 created [Aug 18 10:34:30] DEBUG[15207] stasis.c: Creating topic. name: cache:707/channel:213211, detail: [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15207] stasis.c: Topic 'cache:707/channel:213211': 0x7f0c7c033f60 created [Aug 18 10:34:30] DEBUG[15203] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15199] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15199] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE add candidate: 159.65.48.104:14362, 2130706431 [Aug 18 10:34:30] DEBUG[15199] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15199] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE add candidate: 10.131.0.10:14362, 2130706431 [Aug 18 10:34:30] DEBUG[15199] rtp_engine.c: RTP instance '0x7f0c280a3c30' is setup and ready to go [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) ICE stopped [Aug 18 10:34:30] DEBUG[15199] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15199] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15199] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15199] res_rtp_asterisk.c: (0x7f0c280a3c30) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15199] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15199] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15199] chan_sip.c: SIP call-id changed from '75d72a2f4425b0714f3f80a318268fea@127.0.1.1:5060' to '4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15199] stasis.c: Creating topic. name: channel:213207, detail: [Aug 18 10:34:30] DEBUG[15199] stasis.c: Topic 'channel:213207': 0x7f0c280b0960 created [Aug 18 10:34:30] DEBUG[15199] stasis.c: Creating topic. name: cache:708/channel:213207, detail: [Aug 18 10:34:30] DEBUG[15199] stasis.c: Topic 'cache:708/channel:213207': 0x7f0c280b1390 created [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15213] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15210] res_ari.c: Checking channels externalMedia: Didn't match 213212 [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:476/channel:213153, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:476/channel:213153': 0x7f0c1c0ae9b0 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213153, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213153': 0x7f0c1c091ee0 destroyed [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:30] DEBUG[14988] stasis/app.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15210] res_ari.c: No explicit handler found for 213212. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15214] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15211] http.c: HTTP consuming request body [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213130': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[15215] bridge_channel.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: 0x7f0c08072ae0(Snoop/213006-00000016) is joining [Aug 18 10:34:30] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000c1 - start 1629282863.916031 answer 0.000000 end 1629282870.393142 dur 6.477 bill 1629282870.393 dispo NO ANSWER [Aug 18 10:34:30] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213130' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:414/channel:213130, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:414/channel:213130': 0x7f0cb0079d30 destroyed [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[15043] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:30] DEBUG[15043] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[14741] chan_sip.c: Hangup call SIP/zvonobot-000000b6, SIP callid 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14741] res_rtp_asterisk.c: (0x7f0ca8127f40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14741] res_rtp_asterisk.c: (0x7f0ca8127f40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14741] channel.c: Channel 0x7f0ca811b900 'SIP/zvonobot-000000b6' destroying [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213150': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[13679] channel.c: Channel 0x7f0c9406b7d0 'Recorder/ARI-0000001e;2' destroying [Aug 18 10:34:30] DEBUG[13679] stasis.c: Destroying topic. name: cache:212/channel:1629282839.178, detail: [Aug 18 10:34:30] DEBUG[13679] stasis.c: Topic 'cache:212/channel:1629282839.178': 0x7f0c9405db50 destroyed [Aug 18 10:34:30] DEBUG[13679] stasis.c: Destroying topic. name: channel:1629282839.178, detail: [Aug 18 10:34:30] DEBUG[13679] stasis.c: Topic 'channel:1629282839.178': 0x7f0c9405fd20 destroyed [Aug 18 10:34:30] VERBOSE[13698] app.c: User hung up [Aug 18 10:34:30] DEBUG[13698] res_stasis_recording.c: 1629282838.174: Recording complete [Aug 18 10:34:30] DEBUG[13698] channel.c: Channel 0x7f0c94066640 'Recorder/ARI-0000001e;1' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213150' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Finding handler for bridges/724f7ab9-ed85-4748-9bb7-91218a7c6261/record [Aug 18 10:34:30] DEBUG[14890] channel.c: Channel 0x7f0c940adc90 'Announcer/ARI-0000003f;1' destroying [Aug 18 10:34:30] DEBUG[13678] channel.c: Channel 0x7f0c78059c80 'Recorder/ARI-0000001c;2' destroying [Aug 18 10:34:30] DEBUG[14890] stasis.c: Destroying topic. name: cache:472/channel:1629282858.410, detail: [Aug 18 10:34:30] DEBUG[14890] stasis.c: Topic 'cache:472/channel:1629282858.410': 0x7f0c940b0310 destroyed [Aug 18 10:34:30] DEBUG[14890] stasis.c: Destroying topic. name: channel:1629282858.410, detail: [Aug 18 10:34:30] DEBUG[14890] stasis.c: Topic 'channel:1629282858.410': 0x7f0c94089760 destroyed [Aug 18 10:34:30] DEBUG[15214] http.c: HTTP Request URI is /ari/channels/213016/snoop?app=calls_0&spy=in [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15212] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:30] VERBOSE[15212] bridge_channel.c: Channel Recorder/ARI-0000004d;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:30] DEBUG[13678] stasis.c: Destroying topic. name: cache:210/channel:1629282839.177, detail: [Aug 18 10:34:30] DEBUG[13678] stasis.c: Topic 'cache:210/channel:1629282839.177': 0x7f0c7806aed0 destroyed [Aug 18 10:34:30] DEBUG[13678] stasis.c: Destroying topic. name: channel:1629282839.177, detail: [Aug 18 10:34:30] DEBUG[13678] stasis.c: Topic 'channel:1629282839.177': 0x7f0c78074f80 destroyed [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Finding handler for 724f7ab9-ed85-4748-9bb7-91218a7c6261 [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15213] res_ari.c: No explicit handler found for 724f7ab9-ed85-4748-9bb7-91218a7c6261. Using wildcard bridgeId. [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Finding handler for record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId addChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId removeChannel: Didn't match record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId videoSource: Didn't match record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId moh: Didn't match record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId play: Didn't match record [Aug 18 10:34:30] DEBUG[15213] res_ari.c: Checking bridgeId record: Explicit match with record [Aug 18 10:34:30] DEBUG[15213] stasis.c: Creating topic. name: channel:1629282870.618, detail: [Aug 18 10:34:30] DEBUG[15213] stasis.c: Topic 'channel:1629282870.618': 0x7f0c8807e7d0 created [Aug 18 10:34:30] DEBUG[15213] stasis.c: Creating topic. name: cache:709/channel:1629282870.618, detail: [Aug 18 10:34:30] DEBUG[15213] stasis.c: Topic 'cache:709/channel:1629282870.618': 0x7f0c88063730 created [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 From: ;tag=as7eb98fd0 To: ;tag=as30588395 Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7eb98fd0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as30588395 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: Allocating new SIP dialog for 68c76f653a6ab943067f09237c0567b9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15197] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c20091b50' [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) RTP allocated port 16686 [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE creating session 0.0.0.0:16686 (16686) [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE create [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE add system candidates [Aug 18 10:34:30] DEBUG[15197] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15197] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE add candidate: 159.65.48.104:16686, 2130706431 [Aug 18 10:34:30] DEBUG[15197] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15197] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE add candidate: 10.131.0.10:16686, 2130706431 [Aug 18 10:34:30] DEBUG[15197] rtp_engine.c: RTP instance '0x7f0c20091b50' is setup and ready to go [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) ICE stopped [Aug 18 10:34:30] DEBUG[15214] http.c: match request [ari/channels/213016/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15214] http.c: match request [ari/channels/213016/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15214] http.c: match request [ari/channels/213016/snoop] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15214] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:477/channel:213150, detail: [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: Allocating new SIP dialog for 7f58995f07fa69e46243fd61184b20d7@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15201] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c30047010' [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) RTP allocated port 19488 [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE creating session 0.0.0.0:19488 (19488) [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE create [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE add system candidates [Aug 18 10:34:30] DEBUG[15201] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15201] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:477/channel:213150': 0x7f0cb40320f0 destroyed [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 (Checking To) --From tag as7eb98fd0 --To-tag as30588395 [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE add candidate: 159.65.48.104:19488, 2130706431 [Aug 18 10:34:30] DEBUG[15201] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15201] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE add candidate: 10.131.0.10:19488, 2130706431 [Aug 18 10:34:30] DEBUG[15201] rtp_engine.c: RTP instance '0x7f0c30047010' is setup and ready to go [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) ICE stopped [Aug 18 10:34:30] DEBUG[15201] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15201] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15201] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15201] res_rtp_asterisk.c: (0x7f0c30047010) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15201] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15201] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15201] chan_sip.c: SIP call-id changed from '7f58995f07fa69e46243fd61184b20d7@127.0.1.1:5060' to '7530856d32712b46529332693455f8c4@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15201] stasis.c: Creating topic. name: channel:213209, detail: [Aug 18 10:34:30] DEBUG[15201] stasis.c: Topic 'channel:213209': 0x7f0c300a8840 created [Aug 18 10:34:30] DEBUG[15201] stasis.c: Creating topic. name: cache:710/channel:213209, detail: [Aug 18 10:34:30] DEBUG[15201] stasis.c: Topic 'cache:710/channel:213209': 0x7f0c30169670 created [Aug 18 10:34:30] DEBUG[15197] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.619, detail: [Aug 18 10:34:30] DEBUG[15215] bridge_channel.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: pushing 0x7f0c08072ae0(Snoop/213006-00000016) [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 445 [Aug 18 10:34:30] DEBUG[15197] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 445 [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Finding handler for channels/213213 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Finding handler for channels/213016/snoop [Aug 18 10:34:30] DEBUG[15044] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:30] DEBUG[15044] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[14744] chan_sip.c: Hangup call SIP/zvonobot-000000b7, SIP callid 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14753] chan_sip.c: Hangup call SIP/zvonobot-000000b9, SIP callid 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[15215] bridge_channel.c: Channel Snoop/213006-00000016 joined 'simple_bridge' stasis-bridge <9044ffbe-38f0-4f08-9b7d-173767b0d858> [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.619': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[15052] stasis.c: Creating topic. name: channel:1629282870.621, detail: [Aug 18 10:34:30] DEBUG[15052] stasis.c: Topic 'channel:1629282870.621': 0x7f0c280a4a20 created [Aug 18 10:34:30] DEBUG[15052] stasis.c: Creating topic. name: cache:712/channel:1629282870.621, detail: [Aug 18 10:34:30] DEBUG[15052] stasis.c: Topic 'cache:712/channel:1629282870.621': 0x7f0c280efa90 created [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213130, detail: [Aug 18 10:34:30] DEBUG[14744] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[12982] chan_sip.c: Hangup call SIP/zvonobot-00000014, SIP callid 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14754] chan_sip.c: Hangup call SIP/zvonobot-000000ba, SIP callid 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14754] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:30] DEBUG[14754] res_rtp_asterisk.c: (0x7f0c200c56c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14754] res_rtp_asterisk.c: (0x7f0c200c56c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14754] channel.c: Channel 0x7f0c200cad60 'SIP/zvonobot-000000ba' destroying [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:711/channel:1629282870.619, detail: [Aug 18 10:34:30] DEBUG[14752] chan_sip.c: Hangup call SIP/zvonobot-000000b8, SIP callid 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14752] res_rtp_asterisk.c: (0x7f0c08028a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14752] res_rtp_asterisk.c: (0x7f0c08028a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14744] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14744] channel.c: Channel 0x7f0cb0169950 'SIP/zvonobot-000000b7' destroying [Aug 18 10:34:30] DEBUG[15052] channel.c: Channel 0x7f0c280b5380 'Announcer/ARI-0000005a;1' allocated [Aug 18 10:34:30] DEBUG[15052] stasis.c: Creating topic. name: channel:1629282870.622, detail: [Aug 18 10:34:30] DEBUG[15052] stasis.c: Topic 'channel:1629282870.622': 0x7f0c280cd5d0 created [Aug 18 10:34:30] DEBUG[15052] stasis.c: Creating topic. name: cache:713/channel:1629282870.622, detail: [Aug 18 10:34:30] DEBUG[15052] stasis.c: Topic 'cache:713/channel:1629282870.622': 0x7f0c280f0ca0 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:711/channel:1629282870.619': 0x7f0c300e5e20 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:711/channel:1629282870.619, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:711/channel:1629282870.619': 0x7f0c300e5e20 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.619, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.619': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b6', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213144', '')] [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Finding handler for 213213 [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking channels create: Didn't match 213213 [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15211] res_ari.c: Checking channels externalMedia: Didn't match 213213 [Aug 18 10:34:30] DEBUG[15211] res_ari.c: No explicit handler found for 213213. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[12982] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[12982] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14753] res_rtp_asterisk.c: (0x7f0c9809a4d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14753] res_rtp_asterisk.c: (0x7f0c9809a4d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14753] channel.c: Channel 0x7f0c9808a1b0 'SIP/zvonobot-000000b9' destroying [Aug 18 10:34:30] DEBUG[15197] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213130': 0x7f0cb01270c0 destroyed [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15212] bridge_native_rtp.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4'. Checking compatability for channels 'SIP/zvonobot-00000042' and 'Recorder/ARI-0000004d;2' [Aug 18 10:34:30] DEBUG[14752] channel.c: Channel 0x7f0c08046230 'SIP/zvonobot-000000b8' destroying [Aug 18 10:34:30] DEBUG[12982] channel.c: Channel 0x7f0c24021660 'SIP/zvonobot-00000014' destroying [Aug 18 10:34:30] DEBUG[15197] res_rtp_asterisk.c: (0x7f0c20091b50) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15197] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15197] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[14882] bridge_channel.c: Setting 0x7f0c940b36c0(Announcer/ARI-0000003f;2) state from:0 to:1 [Aug 18 10:34:30] DEBUG[15051] channel.c: Channel 0x7f0c2c08b960 'Recorder/ARI-00000052;1' allocated [Aug 18 10:34:30] DEBUG[14941] channel.c: Channel 0x7f0cac05f500 'Recorder/ARI-00000043;2' destroying [Aug 18 10:34:30] VERBOSE[14963] app.c: User hung up [Aug 18 10:34:30] DEBUG[15051] stasis.c: Creating topic. name: channel:1629282870.623, detail: [Aug 18 10:34:30] DEBUG[15051] stasis.c: Topic 'channel:1629282870.623': 0x7f0c2c087790 created [Aug 18 10:34:30] DEBUG[15051] stasis.c: Creating topic. name: cache:714/channel:1629282870.623, detail: [Aug 18 10:34:30] DEBUG[15051] stasis.c: Topic 'cache:714/channel:1629282870.623': 0x7f0c2c06e960 created [Aug 18 10:34:30] DEBUG[14963] res_stasis_recording.c: 1629282859.434: Recording complete [Aug 18 10:34:30] DEBUG[14963] channel.c: Channel 0x7f0cac02d090 'Recorder/ARI-00000043;1' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[15212] bridge_native_rtp.c: Bridge 'ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4' can not use native RTP bridge as could not get details [Aug 18 10:34:30] DEBUG[14882] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: pulling 0x7f0c940b36c0(Announcer/ARI-0000003f;2) [Aug 18 10:34:30] VERBOSE[14882] bridge_channel.c: Channel Announcer/ARI-0000003f;2 left 'softmix' stasis-bridge [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[14882] bridge_channel.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c940b36c0(Announcer/ARI-0000003f;2) is leaving softmix technology [Aug 18 10:34:30] DEBUG[15212] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213150, detail: [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[13869] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213150': 0x7f0cb4065560 destroyed [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[15062] channel.c: Channel 0x7f0c7c058820 'SIP/zvonobot-000000e6' allocated [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15062] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:30] DEBUG[15212] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[14931] channel.c: Channel 0x7f0c74079320 'Announcer/ARI-00000042;1' destroying [Aug 18 10:34:30] DEBUG[14931] stasis.c: Destroying topic. name: cache:493/channel:1629282858.429, detail: [Aug 18 10:34:30] DEBUG[14931] stasis.c: Topic 'cache:493/channel:1629282858.429': 0x7f0c74049f40 destroyed [Aug 18 10:34:30] DEBUG[14931] stasis.c: Destroying topic. name: channel:1629282858.429, detail: [Aug 18 10:34:30] DEBUG[14931] stasis.c: Topic 'channel:1629282858.429': 0x7f0c740acbb0 destroyed [Aug 18 10:34:30] DEBUG[14917] bridge_channel.c: Setting 0x7f0c740684b0(Announcer/ARI-00000042;2) state from:0 to:1 [Aug 18 10:34:30] DEBUG[14917] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pulling 0x7f0c740684b0(Announcer/ARI-00000042;2) [Aug 18 10:34:30] VERBOSE[14917] bridge_channel.c: Channel Announcer/ARI-00000042;2 left 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:30] DEBUG[14917] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c740684b0(Announcer/ARI-00000042;2) is leaving simple_bridge technology [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[15212] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14917] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[14917] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[14917] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15180] bridge_roles.c: Roles did not exist on channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 [Aug 18 10:34:30] DEBUG[14917] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15215] bridge_native_rtp.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[14917] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[14917] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:30] DEBUG[15212] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213144': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[14941] stasis.c: Destroying topic. name: cache:555/channel:1629282862.482, detail: [Aug 18 10:34:30] DEBUG[15212] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4 is already using the new technology. [Aug 18 10:34:30] DEBUG[14941] stasis.c: Topic 'cache:555/channel:1629282862.482': 0x7f0cac054af0 destroyed [Aug 18 10:34:30] DEBUG[14941] stasis.c: Destroying topic. name: channel:1629282862.482, detail: [Aug 18 10:34:30] DEBUG[14941] stasis.c: Topic 'channel:1629282862.482': 0x7f0cac0822a0 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213144' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[14882] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa'. Checking compatability for channels 'SIP/zvonobot-00000001' and 'Recorder/ARI-0000002c;2' [Aug 18 10:34:30] DEBUG[14882] bridge_native_rtp.c: Bridge 'e8b160c4-f3ae-46ad-bda0-ffa245693ffa' can not use native RTP bridge as channel 'SIP/zvonobot-00000001' has features which prevent it [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[14882] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] VERBOSE[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: switching from softmix technology to simple_bridge [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology constructor [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: moving 0x2c6fb50(SIP/zvonobot-00000001) to dummy bridge temporarily [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: moving 0x7f0c3c10a240(Recorder/ARI-0000002c;2) to dummy bridge temporarily [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is leaving softmix technology (dummy) [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is leaving softmix technology (dummy) [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling softmix technology stop [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x2c6fb50(SIP/zvonobot-00000001) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[14882] channel.c: Channel Recorder/ARI-0000002c;2 setting read format path: slin -> slin [Aug 18 10:34:30] DEBUG[14882] channel.c: Channel Recorder/ARI-0000002c;2 setting write format path: slin -> slin [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: 0x7f0c3c10a240(Recorder/ARI-0000002c;2) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[14882] channel.c: Channel Recorder/ARI-0000002c;2 setting read format path: slin -> slin [Aug 18 10:34:30] DEBUG[14882] channel.c: Channel Recorder/ARI-0000002c;2 setting write format path: slin -> slin [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling simple_bridge technology start [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: deferring softmix technology destructor [Aug 18 10:34:30] DEBUG[14882] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: queueing action type:13 sub:1000 [Aug 18 10:34:30] DEBUG[14917] channel.c: Channel 0x7f0c74046e10 'Announcer/ARI-00000042;2' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[15212] bridge.c: Bridge ad1d4e3d-0b4c-4d73-aef5-268f47efd3a4: 0x7f0c980ad0d0(Recorder/ARI-0000004d;2) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15215] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15065] channel.c: Channel 0x7f0c84065510 'SIP/zvonobot-000000e7' allocated [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15065] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel Recorder/ARI-0000004d;2 setting read format path: slin -> slin [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel SIP/zvonobot-00000042 setting write format path: slin -> ulaw [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel SIP/zvonobot-00000042 setting read format path: ulaw -> slin [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel Recorder/ARI-0000004d;2 setting write format path: slin -> slin [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:478/channel:213144, detail: [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Finding handler for 213016 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channels create: Didn't match 213016 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15215] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15215] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15215] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15215] bridge.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858 is already using the new technology. [Aug 18 10:34:30] DEBUG[15215] bridge.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: 0x7f0c08072ae0(Snoop/213006-00000016) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:478/channel:213144': 0x7f0ca80eb310 destroyed [Aug 18 10:34:30] DEBUG[14988] stasis/app.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' is 2 interested in calls_0 [Aug 18 10:34:30] DEBUG[15204] stasis/control.c: robot_213006: Sending channel add_to_bridge command [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213144, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213144': 0x7f0ca8112590 destroyed [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8127f40) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8127f40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8127f40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca8127f40) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0ca8127f40' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (2) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #146 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #146)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15197] chan_sip.c: SIP call-id changed from '68c76f653a6ab943067f09237c0567b9@127.0.1.1:5060' to '31f6c3a516708506588cba5558cae631@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213145': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channels externalMedia: Didn't match 213016 [Aug 18 10:34:30] DEBUG[15197] stasis.c: Creating topic. name: channel:213208, detail: [Aug 18 10:34:30] DEBUG[15214] res_ari.c: No explicit handler found for 213016. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.625, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.625': 0x7f0c30143340 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:715/channel:1629282870.625, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:715/channel:1629282870.625': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:715/channel:1629282870.625, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:715/channel:1629282870.625': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.625, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.625': 0x7f0c30143340 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b9', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213145', '')] [Aug 18 10:34:30] DEBUG[15065] res_stasis.c: calls_0: Subscribing to 213195 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213145' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:481/channel:213145, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:481/channel:213145': 0x7f0c9802fed0 destroyed [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389;received=159.65.48.104 From: ;tag=as23425771 To: ;tag=as476d86b8 Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35763bfc" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as23425771 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as476d86b8 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35763bfc" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 (Checking To) --From tag as23425771 --To-tag as476d86b8 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Finding handler for snoop [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:30] DEBUG[15197] stasis.c: Topic 'channel:213208': 0x7f0c2009e620 created [Aug 18 10:34:30] DEBUG[15197] stasis.c: Creating topic. name: cache:716/channel:213208, detail: [Aug 18 10:34:30] DEBUG[15197] stasis.c: Topic 'cache:716/channel:213208': 0x7f0c20037450 created [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.626, detail: [Aug 18 10:34:30] DEBUG[15065] stasis/app.c: Channel '213195' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Outgoing Call for 79821116845 [Aug 18 10:34:30] DEBUG[15215] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11 instead [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '212984': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[15065] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15062] res_stasis.c: calls_0: Subscribing to 213194 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '212984' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #68 [Aug 18 10:34:30] DEBUG[15065] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[14311] channel.c: Recorder/ARI-0000002c;2: Dropping redundant connected line update "" <74950493843>. [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[14887] bridge_softmix.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: stopping mixing thread [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:27/channel:212984, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:27/channel:212984': 0x7f0c24023850 destroyed [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:30] DEBUG[20534] bridge.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: calling softmix technology destructor (deferred, dummy) [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:30] DEBUG[20534] bridge_softmix.c: Bridge e8b160c4-f3ae-46ad-bda0-ffa245693ffa: Waiting for mixing thread to die. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '1e454c24705858e9259d323c756ca026@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:30] VERBOSE[15217] chan_sip.c: Audio is at 19406 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] DEBUG[14095] channel.c: SIP/zvonobot-00000001: Dropping redundant connected line update "" <>. [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.626': 0x7f0c30143340 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:717/channel:1629282870.626, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:717/channel:1629282870.626': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:717/channel:1629282870.626, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:717/channel:1629282870.626': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.626, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.626': 0x7f0c30143340 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000ba', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213146', '')] [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389 Max-Forwards: 70 From: ;tag=as23425771 To: ;tag=as476d86b8 Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[15062] stasis/app.c: Channel '213194' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Outgoing Call for 79821116846 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] VERBOSE[15217] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:212984, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.627, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.627': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:718/channel:1629282870.627, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:718/channel:1629282870.627': 0x7f0c3015a290 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:718/channel:1629282870.627, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:718/channel:1629282870.627': 0x7f0c3015a290 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.627, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.627': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:46', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000014', '', 'Stasis', 'calls_0', 42, 9, 'ANSWERED', 3, '', '212984', '')] [Aug 18 10:34:30] VERBOSE[15217] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[14882] channel.c: Channel 0x7f0c940b0650 'Announcer/ARI-0000003f;2' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:212984': 0x7f0c24025020 destroyed [Aug 18 10:34:30] DEBUG[15062] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15062] http.c: HTTP closing session. Top level [Aug 18 10:34:30] VERBOSE[15217] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15073] channel.c: Channel 0x7f0c8c13aeb0 'SIP/zvonobot-000000e8' allocated [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15073] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213146': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213146' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:482/channel:213146, detail: [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:482/channel:213146': 0x7f0c200cd2c0 destroyed [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Initializing initreq for method INVITE - callid 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116845@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 3 [ 52]: From: ;tag=as51a406a7 [Aug 18 10:34:30] DEBUG[14853] res_stasis_recording.c: 1629282864.511: Sending record(213033_cjiiXAUnWZySsSbRJLqGopVrjnYAMqVg.wav) command [Aug 18 10:34:30] DEBUG[15214] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:30] DEBUG[14853] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:30] DEBUG[14853] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 6 [ 60]: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15221] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15220] app.c: play_and_record: , /var/spool/asterisk/recording/213033_cjiiXAUnWZySsSbRJLqGopVrjnYAMqVg, 'wav' [Aug 18 10:34:30] DEBUG[15220] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:30] VERBOSE[15220] app.c: x=0, open writing: /var/spool/asterisk/recording/213033_cjiiXAUnWZySsSbRJLqGopVrjnYAMqVg format: wav, 0x7f0c98006d90 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel Recorder/ARI-0000004d;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 12396 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15212] channel.c: Channel Recorder/ARI-0000004d;2 setting write format path: alaw -> slin [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213149': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.628, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.628': 0x7f0c3015a290 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:719/channel:1629282870.628, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:719/channel:1629282870.628': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:719/channel:1629282870.628, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:719/channel:1629282870.628': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.628, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.628': 0x7f0c3015a290 destroyed [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] VERBOSE[15219] chan_sip.c: Audio is at 19816 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[15221] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #160 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b7', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213149', '')] [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213149' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] DEBUG[15073] res_stasis.c: calls_0: Subscribing to 213201 [Aug 18 10:34:30] VERBOSE[15217] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:479/channel:213149, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:479/channel:213149': 0x7f0cb000bde0 destroyed [Aug 18 10:34:30] DEBUG[15073] stasis/app.c: Channel '213201' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15073] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #59 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[15073] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Aug 18 10:34:30] DEBUG[15217] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[15219] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #59)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] VERBOSE[15217] dial.c: Called zvonobot/79821116845 [Aug 18 10:34:30] DEBUG[15221] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213152': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213152' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: cache:480/channel:213152, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'cache:480/channel:213152': 0x7f0c0806b690 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213152, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213152': 0x7f0c0806b4f0 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213145, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213145': 0x7f0c9808def0 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213146, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213146': 0x7f0c200cd060 destroyed [Aug 18 10:34:30] DEBUG[20620] stasis.c: Destroying topic. name: channel:213149, detail: [Aug 18 10:34:30] DEBUG[20620] stasis.c: Topic 'channel:213149': 0x7f0cb011eee0 destroyed [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '32cbddd62673253d5fac257e298c2963@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c200c56c0) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c200c56c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c200c56c0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c200c56c0) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c200c56c0' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1dc7c43535ddc14e335eb8bd69815b8a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb01036b0) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb01036b0' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '5976fbe32a0b19bd793216aa00518f3b@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9809a4d0) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9809a4d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9809a4d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9809a4d0) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9809a4d0' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c08028a30) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c08028a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c08028a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c08028a30) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c08028a30' [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.629, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.629': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:720/channel:1629282870.629, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:720/channel:1629282870.629': 0x7f0c3015a290 created [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:720/channel:1629282870.629, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:720/channel:1629282870.629': 0x7f0c3015a290 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.629, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.629': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b8', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213152', '')] [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Outgoing Call for 79821116839 [Aug 18 10:34:30] DEBUG[15221] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc;received=159.65.48.104 From: ;tag=as0611ab7b To: ;tag=as096aecc1 Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78bb53e7" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0611ab7b [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as096aecc1 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78bb53e7" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 (Checking To) --From tag as0611ab7b --To-tag as096aecc1 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #72 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc Max-Forwards: 70 From: ;tag=as0611ab7b To: ;tag=as096aecc1 Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[15219] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[15221] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: Allocating new SIP dialog for 3801fb3f2619a2cd605f6f2d5612e67c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15210] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c84062820' [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 12786 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) RTP allocated port 15460 [Aug 18 10:34:30] VERBOSE[15219] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE creating session 0.0.0.0:15460 (15460) [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE create [Aug 18 10:34:30] DEBUG[15221] http.c: Match made with [ari] [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Initializing initreq for method INVITE - callid 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116846@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 3 [ 52]: From: ;tag=as2a65fe21 [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE add system candidates [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Finding handler for bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:30] DEBUG[15221] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:30] DEBUG[15221] stasis.c: Creating topic. name: bridge:4072cacb-4ffa-4a1a-8c8f-2a84e1df6431, detail: [Aug 18 10:34:30] DEBUG[15221] stasis.c: Topic 'bridge:4072cacb-4ffa-4a1a-8c8f-2a84e1df6431': 0x7f0ca0062ee0 created [Aug 18 10:34:30] DEBUG[15221] stasis.c: Creating topic. name: cache:721/bridge:4072cacb-4ffa-4a1a-8c8f-2a84e1df6431, detail: [Aug 18 10:34:30] DEBUG[15221] stasis.c: Topic 'cache:721/bridge:4072cacb-4ffa-4a1a-8c8f-2a84e1df6431': 0x7f0ca00f5680 created [Aug 18 10:34:30] DEBUG[15221] bridge_native_rtp.c: Bridge '4072cacb-4ffa-4a1a-8c8f-2a84e1df6431' can not use native RTP bridge as two channels are required [Aug 18 10:34:30] DEBUG[15221] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] DEBUG[15221] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15210] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15210] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15221] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:30] DEBUG[15221] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15221] bridge.c: Bridge 4072cacb-4ffa-4a1a-8c8f-2a84e1df6431: calling simple_bridge technology constructor [Aug 18 10:34:30] DEBUG[15221] bridge.c: Bridge 4072cacb-4ffa-4a1a-8c8f-2a84e1df6431: calling simple_bridge technology start [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE add candidate: 159.65.48.104:15460, 2130706431 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 6 [ 60]: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] VERBOSE[15219] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #76 [Aug 18 10:34:30] DEBUG[15219] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15210] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15210] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE add candidate: 10.131.0.10:15460, 2130706431 [Aug 18 10:34:30] DEBUG[15210] rtp_engine.c: RTP instance '0x7f0c84062820' is setup and ready to go [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (6) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 157748607 157748607 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16538 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) ICE stopped [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15210] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15210] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15210] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15210] res_rtp_asterisk.c: (0x7f0c84062820) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15210] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15223] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15221] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15223] http.c: HTTP Request URI is /ari/channels/213033/snoop?app=calls_0&spy=in [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #150 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[15210] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: Allocating new SIP dialog for 76ae8a09322416e83e5088df58d7ff6a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15211] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c800a3880' [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) RTP allocated port 15356 [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE creating session 0.0.0.0:15356 (15356) [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE create [Aug 18 10:34:30] DEBUG[15180] stasis/control.c: robot_213006: Adding to bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858 [Aug 18 10:34:30] DEBUG[15180] stasis/app.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' is 3 interested in calls_0 [Aug 18 10:34:30] VERBOSE[15219] dial.c: Called zvonobot/79821116846 [Aug 18 10:34:30] DEBUG[15221] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE add system candidates [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #150)) [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15223] http.c: match request [ari/channels/213033/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15210] chan_sip.c: SIP call-id changed from '3801fb3f2619a2cd605f6f2d5612e67c@127.0.1.1:5060' to '1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15210] stasis.c: Creating topic. name: channel:213212, detail: [Aug 18 10:34:30] DEBUG[15210] stasis.c: Topic 'channel:213212': 0x7f0c84071e10 created [Aug 18 10:34:30] DEBUG[15210] stasis.c: Creating topic. name: cache:722/channel:213212, detail: [Aug 18 10:34:30] DEBUG[15210] stasis.c: Topic 'cache:722/channel:213212': 0x7f0c84072820 created [Aug 18 10:34:30] DEBUG[15223] http.c: match request [ari/channels/213033/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15223] http.c: match request [ari/channels/213033/snoop] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15223] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[15211] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15211] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Finding handler for channels/213033/snoop [Aug 18 10:34:30] DEBUG[15224] bridge_channel.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: 0x7f0cb404b3f0(UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0) is joining [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Finding handler for channels [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c;received=159.65.48.104 From: ;tag=as3f6b0566 To: ;tag=as2be956f9 Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dec0191" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f6b0566 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2be956f9 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dec0191" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 (Checking To) --From tag as3f6b0566 --To-tag as2be956f9 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #48 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] VERBOSE[15222] chan_sip.c: Audio is at 15154 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c Max-Forwards: 70 From: ;tag=as3f6b0566 To: ;tag=as2be956f9 Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 19192 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Finding handler for 213033 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channels create: Didn't match 213033 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channels externalMedia: Didn't match 213033 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: No explicit handler found for 213033. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15224] bridge_channel.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: pushing 0x7f0cb404b3f0(UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0) [Aug 18 10:34:30] VERBOSE[15222] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE add candidate: 159.65.48.104:15356, 2130706431 [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Finding handler for snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:30] VERBOSE[15224] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 joined 'simple_bridge' stasis-bridge <9044ffbe-38f0-4f08-9b7d-173767b0d858> [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:30] DEBUG[15223] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:30] DEBUG[15211] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 - start 1629282870.035257 answer 1629282870.084891 end 1629282870.814662 dur 0.779 bill 0.729 dispo ANSWERED [Aug 18 10:34:30] DEBUG[15211] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15224] bridge_native_rtp.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858'. Checking compatability for channels 'Snoop/213006-00000016' and 'UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:30] DEBUG[15224] bridge_native_rtp.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' can not use native RTP bridge as could not get details [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE add candidate: 10.131.0.10:15356, 2130706431 [Aug 18 10:34:30] DEBUG[15224] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15224] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15211] rtp_engine.c: RTP instance '0x7f0c800a3880' is setup and ready to go [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6387ms with no response [Aug 18 10:34:30] WARNING[20585] chan_sip.c: Hanging up call 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15224] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) ICE stopped [Aug 18 10:34:30] DEBUG[15224] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:30] DEBUG[15224] bridge.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858 is already using the new technology. [Aug 18 10:34:30] DEBUG[15211] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #159 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[15224] bridge.c: Bridge 9044ffbe-38f0-4f08-9b7d-173767b0d858: 0x7f0cb404b3f0(UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0) is joining simple_bridge technology [Aug 18 10:34:30] DEBUG[15211] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #159)) [Aug 18 10:34:30] VERBOSE[15222] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[14858] channel.c: Channel 0x7f0c8411eca0 'SIP/zvonobot-000000c3' hanging up. Refs: 2 [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 setting read format path: slin16 -> slin16 [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel Snoop/213006-00000016 setting write format path: slin16 -> slin [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:30] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000c3 - start 1629282864.079539 answer 0.000000 end 1629282870.821289 dur 6.741 bill 1629282870.821 dispo NO ANSWER [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[15222] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: Allocating new SIP dialog for 30a97943328ba1fa0c84bbb632a67aae@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:30] DEBUG[15205] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c740adb00' [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) RTP allocated port 16394 [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE creating session 0.0.0.0:16394 (16394) [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE create [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE add system candidates [Aug 18 10:34:30] DEBUG[15205] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:30] DEBUG[15205] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE add candidate: 159.65.48.104:16394, 2130706431 [Aug 18 10:34:30] DEBUG[15205] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:30] DEBUG[15205] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE add candidate: 10.131.0.10:16394, 2130706431 [Aug 18 10:34:30] DEBUG[15205] rtp_engine.c: RTP instance '0x7f0c740adb00' is setup and ready to go [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) ICE stopped [Aug 18 10:34:30] DEBUG[15205] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:30] DEBUG[15205] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:30] DEBUG[15205] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15205] res_rtp_asterisk.c: (0x7f0c740adb00) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15205] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel Snoop/213006-00000016 setting read format path: slin -> slin16 [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 setting write format path: slin16 -> slin16 [Aug 18 10:34:30] DEBUG[15211] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:30] DEBUG[15211] res_rtp_asterisk.c: (0x7f0c800a3880) RTCP setup on RTP instance [Aug 18 10:34:30] VERBOSE[15211] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[14671] iostream.c: TCP socket error reading data: Connection reset by peer [Aug 18 10:34:30] DEBUG[14671] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:30] DEBUG[15211] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15204] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:30] DEBUG[15180] stasis/app.c: Bridge '9044ffbe-38f0-4f08-9b7d-173767b0d858' is 4 interested in calls_0 [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 From: ;tag=as22d5765f To: ;tag=as0550790a Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[15204] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK07ba5120;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as22d5765f [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0550790a [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04d570ec" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 (Checking To) --From tag as22d5765f --To-tag as0550790a [Aug 18 10:34:30] DEBUG[15215] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Initializing initreq for method INVITE - callid 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116839@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[15225] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f [Aug 18 10:34:30] DEBUG[15205] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 21dd0eeb71723a880942c3734a0df639@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15211] chan_sip.c: SIP call-id changed from '76ae8a09322416e83e5088df58d7ff6a@127.0.1.1:5060' to '45b0b22714aa16de060380717e05b99e@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #160 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 3 [ 52]: From: ;tag=as3c16b086 [Aug 18 10:34:30] DEBUG[15211] stasis.c: Creating topic. name: channel:213213, detail: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #160)) [Aug 18 10:34:30] DEBUG[15225] http.c: HTTP Request URI is /ari/channels/212974/snoop?app=calls_0&spy=in [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:30] DEBUG[15211] stasis.c: Topic 'channel:213213': 0x7f0c80039960 created [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[15211] stasis.c: Creating topic. name: cache:723/channel:213213, detail: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[15205] chan_sip.c: SIP call-id changed from '30a97943328ba1fa0c84bbb632a67aae@127.0.1.1:5060' to '0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060' [Aug 18 10:34:30] DEBUG[15211] stasis.c: Topic 'cache:723/channel:213213': 0x7f0c80043460 created [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #27)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 6 [ 60]: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] VERBOSE[15222] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c;received=159.65.48.104 From: ;tag=as7e29cf80 To: ;tag=as06f3982c Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="307fff96" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7e29cf80 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as06f3982c [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="307fff96" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 (Checking To) --From tag as7e29cf80 --To-tag as06f3982c [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[15222] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15205] stasis.c: Creating topic. name: channel:213210, detail: [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] DEBUG[15205] stasis.c: Topic 'channel:213210': 0x7f0c740488c0 created [Aug 18 10:34:30] DEBUG[15205] stasis.c: Creating topic. name: cache:724/channel:213210, detail: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[15222] dial.c: Called zvonobot/79821116839 [Aug 18 10:34:30] DEBUG[15225] http.c: match request [ari/channels/212974/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5545fd1c Max-Forwards: 70 From: ;tag=as7e29cf80 To: ;tag=as06f3982c Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15225] http.c: match request [ari/channels/212974/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[15225] http.c: match request [ari/channels/212974/snoop] with handler [ari] len 3 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 10394 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 changing write format from slin16 to slin, native formats (slin16) [Aug 18 10:34:30] DEBUG[15205] stasis.c: Topic 'cache:724/channel:213210': 0x7f0c7401b280 created [Aug 18 10:34:30] DEBUG[15226] http.c: HTTP opening session. Top level [Aug 18 10:34:30] DEBUG[15226] http.c: HTTP Request URI is /ari/channels/robot_212996 [Aug 18 10:34:30] DEBUG[15226] http.c: match request [ari/channels/robot_212996] with handler [httpstatus] len 10 [Aug 18 10:34:30] DEBUG[15226] http.c: match request [ari/channels/robot_212996] with handler [phoneprov] len 9 [Aug 18 10:34:30] DEBUG[15224] channel.c: Channel UnicastRTP/127.0.0.1:50282-0x7f0c200c10c0 setting write format path: slin -> slin16 [Aug 18 10:34:30] DEBUG[15224] res_rtp_asterisk.c: (0x7f0c200c10c0) RTP ooh, format changed from none to slin16 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15226] http.c: match request [ari/channels/robot_212996] with handler [ari] len 3 [Aug 18 10:34:30] DEBUG[15226] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #125 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[15225] http.c: Match made with [ari] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Finding handler for channels/212974/snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Finding handler for channels/robot_212996 [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Finding handler for channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Finding handler for robot_212996 [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking channels create: Didn't match robot_212996 [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15226] res_ari.c: Checking channels externalMedia: Didn't match robot_212996 [Aug 18 10:34:30] DEBUG[15226] res_ari.c: No explicit handler found for robot_212996. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Finding handler for 212974 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channels create: Didn't match 212974 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channels externalMedia: Didn't match 212974 [Aug 18 10:34:30] DEBUG[15105] app.c: One waitfor failed, trying another [Aug 18 10:34:30] DEBUG[15225] res_ari.c: No explicit handler found for 212974. Using wildcard channelId. [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Finding handler for snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:30] DEBUG[15079] channel.c: Channel 0x7f0ca810cd10 'SIP/zvonobot-000000ea' allocated [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15079] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7;received=159.65.48.104 From: ;tag=as06a66f45 To: ;tag=as15b999c2 Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="749a75da" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as06a66f45 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as15b999c2 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="749a75da" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 (Checking To) --From tag as06a66f45 --To-tag as15b999c2 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #100 [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '062e901479878f3469dc381c0f75eb83@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:30] DEBUG[15225] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7 Max-Forwards: 70 From: ;tag=as06a66f45 To: ;tag=as15b999c2 Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[15079] res_stasis.c: calls_0: Subscribing to 213202 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 16822 [Aug 18 10:34:30] DEBUG[15079] stasis/app.c: Channel '213202' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15079] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #50 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (4) INVITE - 5 [Aug 18 10:34:30] DEBUG[15079] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2;received=159.65.48.104 From: ;tag=as109bf1f8 To: ;tag=as04dca485 Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="489df732" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as109bf1f8 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as04dca485 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="489df732" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 (Checking To) --From tag as109bf1f8 --To-tag as04dca485 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '6a9045fb1905dd407eab47186c096641@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a8c5c2 Max-Forwards: 70 From: ;tag=as109bf1f8 To: ;tag=as04dca485 Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Audio is at 13196 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #44 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 From: ;tag=as11b813e8 To: ;tag=as5c9cee45 Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 652472787 652472787 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 19448 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK588f6481;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as11b813e8 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5c9cee45 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:30] DEBUG[15076] channel.c: Channel 0x7f0c88058300 'SIP/zvonobot-000000e9' allocated [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15076] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 652472787 652472787 IN IP4 178.62.121.41 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 19448 RTP/AVP 0 8 101 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Outgoing Call for 79821116838 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 (Checking To) --From tag as11b813e8 --To-tag as5c9cee45 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '08a58ce821a53822322944fb39331df0@159.65.48.104:5060' of Request 103: Match Not Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Strict routing enforced for session 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:30] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:30] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117008@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1d231ba1 Max-Forwards: 70 From: ;tag=as11b813e8 To: ;tag=as5c9cee45 Contact: Call-ID: 08a58ce821a53822322944fb39331df0@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[14095] res_rtp_asterisk.c: (0x7f0cb4008d90) RTCP got report of 100 bytes from 178.62.121.41:14927 [Aug 18 10:34:30] DEBUG[15076] res_stasis.c: calls_0: Subscribing to 213197 [Aug 18 10:34:30] DEBUG[15076] stasis/app.c: Channel '213197' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[15076] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Outgoing Call for 79821116843 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] DEBUG[15076] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[15228] chan_sip.c: Audio is at 14230 [Aug 18 10:34:30] VERBOSE[15228] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[15228] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[15228] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Initializing initreq for method INVITE - callid 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116843@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 3 [ 52]: From: ;tag=as64111725 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34a9f263 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 6 [ 60]: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] VERBOSE[15228] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #69 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:30] DEBUG[15228] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag as34a9f263 [Aug 18 10:34:30] VERBOSE[15228] dial.c: Called zvonobot/79821116843 [Aug 18 10:34:30] VERBOSE[15227] chan_sip.c: Audio is at 12766 [Aug 18 10:34:30] VERBOSE[15227] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[15227] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[15227] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Initializing initreq for method INVITE - callid 104c5130341091f56623cd02618893c9@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116838@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 3 [ 52]: From: ;tag=as768f4f04 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 6 [ 60]: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] VERBOSE[15227] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #152 [Aug 18 10:34:30] DEBUG[15227] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] VERBOSE[15227] dial.c: Called zvonobot/79821116838 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #24 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' of Request 104: Match Found [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (5) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (3) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15077] channel.c: Channel 0x7f0c94046ca0 'SIP/zvonobot-000000eb' allocated [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15077] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] DEBUG[15077] res_stasis.c: calls_0: Subscribing to 213199 [Aug 18 10:34:30] DEBUG[15077] stasis/app.c: Channel '213199' is 1 interested in calls_0 [Aug 18 10:34:30] WARNING[14734] channel.c: Exceptionally long voice queue length queuing to Announcer/ARI-00000038;2 [Aug 18 10:34:30] DEBUG[15077] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Outgoing Call for 79821116841 [Aug 18 10:34:30] DEBUG[15077] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] VERBOSE[15229] chan_sip.c: Audio is at 15450 [Aug 18 10:34:30] DEBUG[14766] chan_sip.c: Hangup call SIP/zvonobot-000000bc, SIP callid 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:30] VERBOSE[15229] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] DEBUG[14766] res_rtp_asterisk.c: (0x7f0c180d82e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[14766] res_rtp_asterisk.c: (0x7f0c180d82e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[15078] channel.c: Channel 0x7f0c900dfbe0 'SIP/zvonobot-000000ec' allocated [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:30] DEBUG[15078] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:30] VERBOSE[15229] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389;received=159.65.48.104 From: ;tag=as23425771 To: ;tag=as476d86b8 Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35763bfc" Content-Length: 0 <-------------> [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:30] DEBUG[14766] channel.c: Channel 0x7f0c18105090 'SIP/zvonobot-000000bc' destroying [Aug 18 10:34:30] VERBOSE[15229] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK41367389;received=159.65.48.104 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as23425771 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as476d86b8 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Initializing initreq for method INVITE - callid 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116841@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35763bfc" [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:30] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213148': is 0 interested in calls_0 [Aug 18 10:34:30] DEBUG[20620] stasis/app.c: channel '213148' unsubscribed from calls_0 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282870.633, detail: [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 3 [ 52]: From: ;tag=as70796578 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.633': 0x7f0c300adb10 created [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 (Checking To) --From tag as23425771 --To-tag as476d86b8 [Aug 18 10:34:30] DEBUG[20545] stasis.c: Creating topic. name: cache:725/channel:1629282870.633, detail: [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Stopping retransmission on '1e454c24705858e9259d323c756ca026@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:725/channel:1629282870.633': 0x7f0c300117a0 created [Aug 18 10:34:30] DEBUG[14766] stasis.c: Destroying topic. name: cache:488/channel:213148, detail: [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[14766] stasis.c: Topic 'cache:488/channel:213148': 0x7f0c180fef10 destroyed [Aug 18 10:34:30] DEBUG[14766] stasis.c: Destroying topic. name: channel:213148, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: cache:725/channel:1629282870.633, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'cache:725/channel:1629282870.633': 0x7f0c300117a0 destroyed [Aug 18 10:34:30] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282870.633, detail: [Aug 18 10:34:30] DEBUG[20545] stasis.c: Topic 'channel:1629282870.633': 0x7f0c300adb10 destroyed [Aug 18 10:34:30] DEBUG[14766] stasis.c: Topic 'channel:213148': 0x7f0c18099850 destroyed [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000bc', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213148', '')] [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 6 [ 60]: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:30] DEBUG[15078] res_stasis.c: calls_0: Subscribing to 213200 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #23 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[15078] stasis/app.c: Channel '213200' is 1 interested in calls_0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #23)) [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Outgoing Call for 79821116840 [Aug 18 10:34:30] DEBUG[15078] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #125 (1) INVITE - 5 [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #125)) [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:30] DEBUG[20585] chan_sip.c: Destroying SIP dialog 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:30] VERBOSE[15230] chan_sip.c: Audio is at 10424 [Aug 18 10:34:30] VERBOSE[15229] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:30] VERBOSE[15230] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:30] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #67 [Aug 18 10:34:30] DEBUG[15229] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:30] VERBOSE[15230] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS stop [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] VERBOSE[15230] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:30] DEBUG[15078] http.c: HTTP closing session. Top level [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) DTLS srtp - stopped timeout timer' [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:30] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c7c020d90) ICE RTP transport deallocating [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Initializing initreq for method INVITE - callid 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116840@178.62.121.41 SIP/2.0 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:30] VERBOSE[15229] dial.c: Called zvonobot/79821116841 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 3 [ 52]: From: ;tag=as6ca8b41e [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 6 [ 60]: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 [Aug 18 10:34:30] DEBUG[15230] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:30 GMT [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] VERBOSE[15230] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #72 [Aug 18 10:34:31] DEBUG[15230] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c7c020d90' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:31] VERBOSE[15230] dial.c: Called zvonobot/79821116840 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[14716] channel.c: Channel 0x7f0c200b0230 'Recorder/ARI-00000039;2' destroying [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15086] channel.c: Channel 0x7f0ca403d4d0 'Announcer/ARI-00000054;1' allocated [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15086] stasis.c: Creating topic. name: channel:1629282871.634, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15086] stasis.c: Topic 'channel:1629282871.634': 0x7f0ca410a4b0 created [Aug 18 10:34:31] DEBUG[14716] stasis.c: Destroying topic. name: cache:461/channel:1629282857.401, detail: [Aug 18 10:34:31] DEBUG[14716] stasis.c: Topic 'cache:461/channel:1629282857.401': 0x7f0c20085f10 destroyed [Aug 18 10:34:31] DEBUG[14716] stasis.c: Destroying topic. name: channel:1629282857.401, detail: [Aug 18 10:34:31] DEBUG[14716] stasis.c: Topic 'channel:1629282857.401': 0x7f0c200b10f0 destroyed [Aug 18 10:34:31] DEBUG[15086] stasis.c: Creating topic. name: cache:726/channel:1629282871.634, detail: [Aug 18 10:34:31] DEBUG[13561] res_rtp_asterisk.c: (0x7f0c8c06a190) RTP 0x7f0c8c06ef60 -- Received packet from 178.62.121.41:10896, dropping due to strict RTP protection. [Aug 18 10:34:31] DEBUG[13569] res_rtp_asterisk.c: (0x7f0cac04dff0) RTP 0x7f0cac04fb10 -- Received packet from 178.62.121.41:17066, dropping due to strict RTP protection. [Aug 18 10:34:31] DEBUG[15086] stasis.c: Topic 'cache:726/channel:1629282871.634': 0x7f0ca4108e90 created [Aug 18 10:34:31] DEBUG[13132] res_rtp_asterisk.c: (0x7f0ca401ab20) RTP 0x7f0ca401e710 -- Received packet from 178.62.121.41:14754, dropping due to strict RTP protection. [Aug 18 10:34:31] DEBUG[15089] channel.c: Channel 0x7f0cb0112830 'UnicastRTP/127.0.0.1:50456-0x7f0cb014c030' allocated [Aug 18 10:34:31] DEBUG[15089] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:31] VERBOSE[15089] res_rtp_asterisk.c: 0x7f0cb0114280 -- Strict RTP learning after remote address set to: 127.0.0.1:50456 [Aug 18 10:34:31] DEBUG[15089] res_stasis.c: calls_0: Subscribing to robot_213036 [Aug 18 10:34:31] DEBUG[15089] stasis/app.c: Channel 'robot_213036' is 1 interested in calls_0 [Aug 18 10:34:31] VERBOSE[15231] dial.c: Called 127.0.0.1:50456 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:31] VERBOSE[15231] dial.c: UnicastRTP/127.0.0.1:50456-0x7f0cb014c030 answered [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:31] VERBOSE[15231] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50456-0x7f0cb014c030 [Aug 18 10:34:31] DEBUG[15089] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15089] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:31] DEBUG[15231] stasis/app.c: Channel 'robot_213036' is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[14519] res_rtp_asterisk.c: (0x7f0c980a1440) RTCP got report of 100 bytes from 178.62.121.41:12381 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK442c1a3c Max-Forwards: 70 From: ;tag=as342fd06d To: ;tag=as18b114f0 Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="215d54e3", response="0f0f485dd16f4c0cc7ced26f37245ffb" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK442c1a3c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as342fd06d [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as18b114f0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="215d54e3", response="0f0f485dd16f4c0cc7ced26f37245ffb" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 (Checking From) --From tag as342fd06d --To-tag as18b114f0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15080] channel.c: Channel 0x7f0c9c0eec40 'SIP/zvonobot-000000ed' allocated [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[15080] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c20028ba0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[15080] res_stasis.c: calls_0: Subscribing to 213203 [Aug 18 10:34:31] DEBUG[15080] stasis/app.c: Channel '213203' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[14889] stasis.c: Creating topic. name: channel:1629282871.635, detail: [Aug 18 10:34:31] DEBUG[14889] stasis.c: Topic 'channel:1629282871.635': 0x7f0c38041170 created [Aug 18 10:34:31] DEBUG[14889] stasis.c: Creating topic. name: cache:727/channel:1629282871.635, detail: [Aug 18 10:34:31] DEBUG[14889] stasis.c: Topic 'cache:727/channel:1629282871.635': 0x7f0c38036c60 created [Aug 18 10:34:31] DEBUG[15080] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15080] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Outgoing Call for 79821116837 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[15232] chan_sip.c: Audio is at 12398 [Aug 18 10:34:31] VERBOSE[15232] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] VERBOSE[15232] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] VERBOSE[15232] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '549831e20d284c8653320b2042dceffc@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK442c1a3c;received=178.62.121.41 From: ;tag=as342fd06d To: ;tag=as18b114f0 Call-ID: 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Initializing initreq for method INVITE - callid 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116837@178.62.121.41 SIP/2.0 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Setting 0x7f0c3017b4a0(SIP/zvonobot-00000034) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 3 [ 52]: From: ;tag=as6760f146 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 6 [ 60]: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:31 GMT [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] VERBOSE[15232] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #96 [Aug 18 10:34:31] DEBUG[15232] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[15232] dial.c: Called zvonobot/79821116837 [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: pulling 0x7f0c3017b4a0(SIP/zvonobot-00000034) [Aug 18 10:34:31] VERBOSE[14810] bridge_channel.c: Channel SIP/zvonobot-00000034 left 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c3017b4a0(SIP/zvonobot-00000034) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Setting 0x7f0c0805d570(Recorder/ARI-0000004c;2) state from:0 to:2 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:31] DEBUG[14810] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14810] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14810] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14810] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14810] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[14810] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad is already using the new technology. [Aug 18 10:34:31] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000034 - start 1629282832.108777 answer 1629282862.942044 end 1629282871.088893 dur 38.980 bill 8.146 dispo ANSWERED [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15174] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: pulling 0x7f0c0805d570(Recorder/ARI-0000004c;2) [Aug 18 10:34:31] VERBOSE[15174] bridge_channel.c: Channel Recorder/ARI-0000004c;2 left 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[15174] bridge_channel.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad: 0x7f0c0805d570(Recorder/ARI-0000004c;2) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Bridge is returning 0x7f0c3017b4a0(SIP/zvonobot-00000034) to read format alaw [Aug 18 10:34:31] DEBUG[15174] bridge_native_rtp.c: Bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[15174] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14810] channel.c: Channel SIP/zvonobot-00000034 setting read format path: alaw -> alaw [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #142 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[15174] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #142)) [Aug 18 10:34:31] DEBUG[14810] bridge_channel.c: Bridge is returning 0x7f0c3017b4a0(SIP/zvonobot-00000034) to write format alaw [Aug 18 10:34:31] DEBUG[15174] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14810] channel.c: Channel SIP/zvonobot-00000034 setting write format path: alaw -> alaw [Aug 18 10:34:31] DEBUG[14810] stasis/control.c: 213016, f58763a3-c201-4609-b9b6-f8cb14b257ad: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[15174] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14810] stasis/app.c: bridge 'f58763a3-c201-4609-b9b6-f8cb14b257ad': is 1 interested in calls_0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15174] bridge.c: Bridge f58763a3-c201-4609-b9b6-f8cb14b257ad is already using the new technology. [Aug 18 10:34:31] DEBUG[14810] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13271] stasis/control.c: 213016: Channel departing bridge [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[13271] bridge.c: Waiting for 0x7f0c3017b4a0(SIP/zvonobot-00000034) bridge thread to die. [Aug 18 10:34:31] DEBUG[13271] stasis/app.c: channel '213016': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[13271] channel.c: Channel 0x7f0c20031440 'SIP/zvonobot-00000034' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:31] DEBUG[15174] channel.c: Channel 0x7f0c0807aab0 'Recorder/ARI-0000004c;2' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #160 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #160)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #152 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[15068] channel.c: Channel 0x7f0c80033cc0 'SIP/zvonobot-000000ee' allocated [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #152)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4001bc1324c663963b99d0721ed3598c@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180d82e0) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180d82e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180d82e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c180d82e0) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c180d82e0' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Session timer stopped: 95 - 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661387 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15068] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0;received=159.65.48.104 From: ;tag=as293daefc To: ;tag=as38878219 Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e75e235" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as293daefc [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as38878219 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e75e235" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 (Checking To) --From tag as293daefc --To-tag as38878219 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 [Aug 18 10:34:31] DEBUG[14779] chan_sip.c: Hangup call SIP/zvonobot-000000bb, SIP callid 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[14779] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:31] DEBUG[14884] channel.c: Channel 0x7f0c7c0d8220 'Recorder/ARI-0000003e;1' destroying [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:31] DEBUG[14779] res_rtp_asterisk.c: (0x2c96ae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14779] res_rtp_asterisk.c: (0x2c96ae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14779] channel.c: Channel 0x2c9e500 'SIP/zvonobot-000000bb' destroying [Aug 18 10:34:31] DEBUG[14770] channel.c: Channel 0x7f0c080937a0 'Recorder/ARI-00000048;2' allocated [Aug 18 10:34:31] DEBUG[14770] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[15233] bridge_channel.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: 0x7f0c0800e860(Recorder/ARI-00000048;2) is joining [Aug 18 10:34:31] DEBUG[15233] bridge_channel.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: pushing 0x7f0c0800e860(Recorder/ARI-00000048;2) [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:31] DEBUG[14884] stasis.c: Destroying topic. name: cache:470/channel:1629282858.407, detail: [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK34eae5c0 Max-Forwards: 70 From: ;tag=as293daefc To: ;tag=as38878219 Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:31] DEBUG[14884] stasis.c: Topic 'cache:470/channel:1629282858.407': 0x7f0c7c087cf0 destroyed [Aug 18 10:34:31] DEBUG[14884] stasis.c: Destroying topic. name: channel:1629282858.407, detail: [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213151': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213151' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14884] stasis.c: Topic 'channel:1629282858.407': 0x7f0c7c0a5470 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.636, detail: [Aug 18 10:34:31] DEBUG[15233] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[15233] bridge_channel.c: Channel Recorder/ARI-00000048;2 joined 'simple_bridge' stasis-bridge <1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be> [Aug 18 10:34:31] DEBUG[14779] stasis.c: Destroying topic. name: cache:483/channel:213151, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:31] DEBUG[14779] stasis.c: Topic 'cache:483/channel:213151': 0x2c230c0 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.636': 0x7f0c300e9460 created [Aug 18 10:34:31] DEBUG[14779] stasis.c: Destroying topic. name: channel:213151, detail: [Aug 18 10:34:31] DEBUG[14779] stasis.c: Topic 'channel:213151': 0x2c23b60 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:728/channel:1629282871.636, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:728/channel:1629282871.636': 0x7f0c300200e0 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[15081] channel.c: Channel 0x7f0c980b3f80 'SIP/zvonobot-000000ef' allocated [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Audio is at 13406 [Aug 18 10:34:31] DEBUG[15233] bridge_native_rtp.c: Bridge '1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be'. Checking compatability for channels 'SIP/zvonobot-0000000d' and 'Recorder/ARI-00000048;2' [Aug 18 10:34:31] DEBUG[15233] bridge_native_rtp.c: Bridge '1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be' can not use native RTP bridge as could not get details [Aug 18 10:34:31] DEBUG[15233] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:728/channel:1629282871.636, detail: [Aug 18 10:34:31] DEBUG[15233] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:728/channel:1629282871.636': 0x7f0c300200e0 destroyed [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[15233] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15081] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[15233] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] DEBUG[15233] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be is already using the new technology. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] DEBUG[15068] res_stasis.c: calls_0: Subscribing to 213196 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.636, detail: [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.636': 0x7f0c300e9460 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000bb', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213151', '')] [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15233] bridge.c: Bridge 1af6b0f0-8ce9-44ee-8d82-dc8e2b9f74be: 0x7f0c0800e860(Recorder/ARI-00000048;2) is joining simple_bridge technology [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel Recorder/ARI-00000048;2 setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[15081] res_stasis.c: calls_0: Subscribing to 213198 [Aug 18 10:34:31] DEBUG[15068] stasis/app.c: Channel '213196' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15081] stasis/app.c: Channel '213198' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel SIP/zvonobot-0000000d setting write format path: slin -> ulaw [Aug 18 10:34:31] DEBUG[15081] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Outgoing Call for 79821116844 [Aug 18 10:34:31] DEBUG[15081] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel SIP/zvonobot-0000000d setting read format path: ulaw -> slin [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel Recorder/ARI-00000048;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[15234] chan_sip.c: Audio is at 14858 [Aug 18 10:34:31] VERBOSE[15234] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:31] DEBUG[15068] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15068] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Outgoing Call for 79821116842 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] VERBOSE[15234] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #153 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #153)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #23 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #23)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #155 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #155)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #125 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #125)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #157 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #157)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[15234] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Initializing initreq for method INVITE - callid 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8;received=159.65.48.104 From: ;tag=as7a3cd0ea To: ;tag=as3e98a678 Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4676557d" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116844@178.62.121.41 SIP/2.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7eabbca8;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7a3cd0ea [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 3 [ 52]: From: ;tag=as7417feac [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3e98a678 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 6 [ 60]: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4676557d" [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:31 GMT [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 (Checking To) --From tag as7a3cd0ea --To-tag as3e98a678 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 769f001c396d5c3427cb42413aa68f9b@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[15234] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #110 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '0472acf14bdf1d08522854a959df46d6@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x2c96ae0) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x2c96ae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x2c96ae0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x2c96ae0) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[15234] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x2c96ae0' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:31] DEBUG[14770] res_stasis_recording.c: 1629282862.471: Sending record(212979_JmGFaAKGpLdhTmasRWipwagxgDaiCpdz.wav) command [Aug 18 10:34:31] VERBOSE[15234] dial.c: Called zvonobot/79821116844 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14894] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14894] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] DEBUG[14770] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel Recorder/ARI-00000048;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:31] DEBUG[15233] channel.c: Channel Recorder/ARI-00000048;2 setting write format path: alaw -> slin [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936;received=159.65.48.104 From: ;tag=as73a421e4 To: ;tag=as6af305f3 Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59ecf5fc" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as73a421e4 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6af305f3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] VERBOSE[15235] chan_sip.c: Audio is at 11540 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59ecf5fc" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 (Checking To) --From tag as73a421e4 --To-tag as6af305f3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:31] VERBOSE[15235] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5d877936 Max-Forwards: 70 From: ;tag=as73a421e4 To: ;tag=as6af305f3 Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:31] VERBOSE[15235] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Audio is at 17014 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #88 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15239] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[14770] http.c: HTTP closing session. Top level [Aug 18 10:34:31] VERBOSE[15235] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Initializing initreq for method INVITE - callid 46be811217bc41126929634752a2647e@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116842@178.62.121.41 SIP/2.0 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 3 [ 52]: From: ;tag=as539445e1 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 6 [ 60]: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:31 GMT [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] VERBOSE[15235] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae;received=159.65.48.104 From: ;tag=as406d4539 To: ;tag=as27a3d6f4 Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="222035a3" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as406d4539 [Aug 18 10:34:31] DEBUG[15239] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as27a3d6f4 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #139 [Aug 18 10:34:31] DEBUG[15235] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="222035a3" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 (Checking To) --From tag as406d4539 --To-tag as27a3d6f4 [Aug 18 10:34:31] DEBUG[15239] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #78 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae Max-Forwards: 70 From: ;tag=as406d4539 To: ;tag=as27a3d6f4 Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Audio is at 14666 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15239] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15239] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15239] http.c: Match made with [ari] [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call leg/transaction does not exist [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15239] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15239] stasis.c: Creating topic. name: bridge:a125ffe7-7c6f-40c7-9445-d81806616772, detail: [Aug 18 10:34:31] DEBUG[15239] stasis.c: Topic 'bridge:a125ffe7-7c6f-40c7-9445-d81806616772': 0x7f0c3c00a510 created [Aug 18 10:34:31] DEBUG[15239] stasis.c: Creating topic. name: cache:729/bridge:a125ffe7-7c6f-40c7-9445-d81806616772, detail: [Aug 18 10:34:31] DEBUG[15239] stasis.c: Topic 'cache:729/bridge:a125ffe7-7c6f-40c7-9445-d81806616772': 0x7f0c3c006fa0 created [Aug 18 10:34:31] DEBUG[15239] bridge_native_rtp.c: Bridge 'a125ffe7-7c6f-40c7-9445-d81806616772' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[15239] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15239] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15239] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:31] DEBUG[15239] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15239] bridge.c: Bridge a125ffe7-7c6f-40c7-9445-d81806616772: calling simple_bridge technology constructor [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] DEBUG[15239] bridge.c: Bridge a125ffe7-7c6f-40c7-9445-d81806616772: calling simple_bridge technology start [Aug 18 10:34:31] DEBUG[15236] app.c: play_and_record: , /var/spool/asterisk/recording/212979_JmGFaAKGpLdhTmasRWipwagxgDaiCpdz, 'wav' [Aug 18 10:34:31] DEBUG[15236] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34a9f263 [Aug 18 10:34:31] DEBUG[14897] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14897] http.c: HTTP closing session. Top level [Aug 18 10:34:31] VERBOSE[15235] dial.c: Called zvonobot/79821116842 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:31] DEBUG[14762] res_rtp_asterisk.c: (0x7f0c90008240) RTCP got report of 76 bytes from 178.62.121.41:12819 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag as34a9f263 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 474 [Aug 18 10:34:31] VERBOSE[15236] app.c: x=0, open writing: /var/spool/asterisk/recording/212979_JmGFaAKGpLdhTmasRWipwagxgDaiCpdz format: wav, 0x7f0c30177130 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #152 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #152)) [Aug 18 10:34:31] DEBUG[15239] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15239] http.c: HTTP closing session. Top level [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTCP got report of 100 bytes from 178.62.121.41:10695 [Aug 18 10:34:31] DEBUG[15248] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15248] http.c: HTTP Request URI is /ari/channels/212979/snoop?app=calls_0&spy=in [Aug 18 10:34:31] DEBUG[14898] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14898] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15248] http.c: match request [ari/channels/212979/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15248] http.c: match request [ari/channels/212979/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15248] http.c: match request [ari/channels/212979/snoop] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15248] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[13141] chan_sip.c: Hangup call SIP/zvonobot-00000027, SIP callid 25fe09dd2e48bced380a74a44e7195f3@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[13141] res_rtp_asterisk.c: (0x7f0c0801b610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[13141] res_rtp_asterisk.c: (0x7f0c0801b610) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Finding handler for channels/212979/snoop [Aug 18 10:34:31] DEBUG[14896] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14878] channel.c: Channel 0x7f0c7c054de0 'Recorder/ARI-0000003e;2' destroying [Aug 18 10:34:31] DEBUG[14896] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[13201] chan_sip.c: Hangup call SIP/zvonobot-00000030, SIP callid 710394295318048c14806fba23b501f2@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[13201] res_rtp_asterisk.c: (0x7f0c9801aec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[13201] res_rtp_asterisk.c: (0x7f0c9801aec0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14639] chan_sip.c: Hangup call SIP/zvonobot-000000af, SIP callid 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[14639] res_rtp_asterisk.c: (0x7f0c080681f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14639] res_rtp_asterisk.c: (0x7f0c080681f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14639] channel.c: Channel 0x7f0c080f4050 'SIP/zvonobot-000000af' destroying [Aug 18 10:34:31] DEBUG[13632] channel.c: Channel 0x7f0c7c07a200 'Recorder/ARI-00000019;1' destroying [Aug 18 10:34:31] DEBUG[13632] stasis.c: Destroying topic. name: cache:182/channel:1629282837.153, detail: [Aug 18 10:34:31] DEBUG[13632] stasis.c: Topic 'cache:182/channel:1629282837.153': 0x7f0c7c07cf30 destroyed [Aug 18 10:34:31] DEBUG[13632] stasis.c: Destroying topic. name: channel:1629282837.153, detail: [Aug 18 10:34:31] DEBUG[13632] stasis.c: Topic 'channel:1629282837.153': 0x7f0c7c02e7e0 destroyed [Aug 18 10:34:31] DEBUG[14617] channel.c: Channel 0x7f0c24011df0 'SIP/zvonobot-00000004' destroying [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[14734] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:31] DEBUG[14047] bridge_channel.c: Setting 0x7f0c080f3ea0(Snoop/213011-0000000f) state from:0 to:1 [Aug 18 10:34:31] DEBUG[14734] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:31] DEBUG[14878] stasis.c: Destroying topic. name: cache:529/channel:1629282860.459, detail: [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:31] DEBUG[14715] channel.c: Channel 0x7f0c2412f860 'Announcer/ARI-00000038;2' destroying [Aug 18 10:34:31] DEBUG[14734] channel.c: Channel Announcer/ARI-00000038;1 setting write format path: slin -> slin [Aug 18 10:34:31] WARNING[14734] res_stasis_playback.c: 1629282854.363: Playback failed for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:31] DEBUG[14734] res_stasis_playback.c: Channel: Announcer/ARI-00000038;1 already hangup, stop playback [Aug 18 10:34:31] NOTICE[14734] res_stasis_playback.c: 1629282854.363: Playback canceled for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213140': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[14715] stasis.c: Destroying topic. name: cache:462/channel:1629282857.402, detail: [Aug 18 10:34:31] DEBUG[14715] stasis.c: Topic 'cache:462/channel:1629282857.402': 0x7f0c24131220 destroyed [Aug 18 10:34:31] DEBUG[14715] stasis.c: Destroying topic. name: channel:1629282857.402, detail: [Aug 18 10:34:31] DEBUG[14715] stasis.c: Topic 'channel:1629282857.402': 0x7f0c24130800 destroyed [Aug 18 10:34:31] DEBUG[14878] stasis.c: Topic 'cache:529/channel:1629282860.459': 0x7f0c7c087b60 destroyed [Aug 18 10:34:31] DEBUG[14878] stasis.c: Destroying topic. name: channel:1629282860.459, detail: [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213140' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14734] channel.c: Channel 0x7f0c2419f2d0 'Announcer/ARI-00000038;1' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[14047] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: pulling 0x7f0c080f3ea0(Snoop/213011-0000000f) [Aug 18 10:34:31] VERBOSE[14047] bridge_channel.c: Channel Snoop/213011-0000000f left 'simple_bridge' stasis-bridge <28c87384-44a9-4ebc-9328-4118df068e33> [Aug 18 10:34:31] DEBUG[14047] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c080f3ea0(Snoop/213011-0000000f) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[14878] stasis.c: Topic 'channel:1629282860.459': 0x7f0c7c0a73e0 destroyed [Aug 18 10:34:31] DEBUG[14639] stasis.c: Destroying topic. name: cache:440/channel:213140, detail: [Aug 18 10:34:31] DEBUG[14639] stasis.c: Topic 'cache:440/channel:213140': 0x7f0c080772b0 destroyed [Aug 18 10:34:31] DEBUG[14639] stasis.c: Destroying topic. name: channel:213140, detail: [Aug 18 10:34:31] DEBUG[14639] stasis.c: Topic 'channel:213140': 0x7f0c08041100 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.637, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.637': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:730/channel:1629282871.637, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:730/channel:1629282871.637': 0x7f0c300adb10 created [Aug 18 10:34:31] DEBUG[14047] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14047] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:31] DEBUG[14047] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14047] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14047] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14047] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33 is already using the new technology. [Aug 18 10:34:31] DEBUG[14617] channel.c: Channel 0x7f0c880cb930 'Snoop/212967-00000015' destroying [Aug 18 10:34:31] DEBUG[14617] stasis.c: Destroying topic. name: cache:487/channel:1629282858.425, detail: [Aug 18 10:34:31] DEBUG[14617] stasis.c: Topic 'cache:487/channel:1629282858.425': 0x7f0c880877e0 destroyed [Aug 18 10:34:31] DEBUG[14617] stasis.c: Destroying topic. name: channel:1629282858.425, detail: [Aug 18 10:34:31] DEBUG[14617] stasis.c: Topic 'channel:1629282858.425': 0x7f0c880590d0 destroyed [Aug 18 10:34:31] DEBUG[14047] bridge_channel.c: Bridge is returning 0x7f0c080f3ea0(Snoop/213011-0000000f) to read format slin [Aug 18 10:34:31] DEBUG[14047] channel.c: Channel Snoop/213011-0000000f setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[14047] bridge_channel.c: Bridge is returning 0x7f0c080f3ea0(Snoop/213011-0000000f) to write format slin [Aug 18 10:34:31] DEBUG[14047] channel.c: Channel Snoop/213011-0000000f setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:31] DEBUG[14047] stasis/control.c: 1629282842.223, 28c87384-44a9-4ebc-9328-4118df068e33: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:31] DEBUG[14047] stasis/app.c: bridge '28c87384-44a9-4ebc-9328-4118df068e33': is 3 interested in calls_0 [Aug 18 10:34:31] DEBUG[14047] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13896] stasis/control.c: 1629282842.223: Channel departing bridge [Aug 18 10:34:31] DEBUG[13896] bridge.c: Waiting for 0x7f0c080f3ea0(Snoop/213011-0000000f) bridge thread to die. [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212967': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212967' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:730/channel:1629282871.637, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:730/channel:1629282871.637': 0x7f0c300adb10 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.637, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.637': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:11/channel:212967, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:11/channel:212967': 0x7f0c24078350 destroyed [Aug 18 10:34:31] DEBUG[13896] stasis/app.c: channel '1629282842.223': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[13896] stasis/app.c: channel '1629282842.223' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[13896] channel.c: Channel 0x7f0c2c0b7210 'Snoop/213011-0000000f' hanging up. Refs: 3 [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000af', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213140', '')] [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[14643] channel.c: Channel 0x2c2aeb0 'Recorder/ARI-00000044;2' allocated [Aug 18 10:34:31] DEBUG[14643] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:212967, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:212967': 0x7f0c24078190 destroyed [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.638, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.638': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:731/channel:1629282871.638, detail: [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Finding handler for 212979 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channels create: Didn't match 212979 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[14919] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[15249] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x23a5040(Recorder/ARI-00000044;2) is joining [Aug 18 10:34:31] DEBUG[14919] http.c: HTTP closing session. Top level [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424 Max-Forwards: 70 From: ;tag=as0d3ccf68 To: ;tag=as4d3d785f Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="29c76a44", response="e2ffe165291edb147174178f90098442" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:731/channel:1629282871.638': 0x7f0c300adb10 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channels externalMedia: Didn't match 212979 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as0d3ccf68 [Aug 18 10:34:31] DEBUG[15249] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: pushing 0x23a5040(Recorder/ARI-00000044;2) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as4d3d785f [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="29c76a44", response="e2ffe165291edb147174178f90098442" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 (Checking From) --From tag as0d3ccf68 --To-tag as4d3d785f [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 102, ours 102) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Received bye, no owner, selfdestruct soon. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK1f5eb424;received=178.62.121.41 From: ;tag=as0d3ccf68 To: ;tag=as4d3d785f Call-ID: 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14927] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14927] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15248] res_ari.c: No explicit handler found for 212979. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:731/channel:1629282871.638, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:731/channel:1629282871.638': 0x7f0c300adb10 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.638, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.638': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:42', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000004', '', 'Stasis', 'calls_0', 43, 18, 'ANSWERED', 3, '', '212967', '')] [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Finding handler for snoop [Aug 18 10:34:31] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6118ms with no response [Aug 18 10:34:31] WARNING[20585] chan_sip.c: Hanging up call 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15249] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] VERBOSE[15249] bridge_channel.c: Channel Recorder/ARI-00000044;2 joined 'simple_bridge' stasis-bridge <724f7ab9-ed85-4748-9bb7-91218a7c6261> [Aug 18 10:34:31] DEBUG[14928] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20545] cdr.c: Finalized CDR for Snoop/213011-0000000f - start 1629282843.388036 answer 1629282843.388036 end 1629282871.283676 dur 27.895 bill 27.895 dispo ANSWERED [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.639, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14928] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2188851b40d9c7be5b3642177dd5d9ec@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c080681f0) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c080681f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c080681f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c080681f0) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c080681f0' [Aug 18 10:34:31] DEBUG[15249] bridge_native_rtp.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261'. Checking compatability for channels 'SIP/zvonobot-0000003d' and 'Recorder/ARI-00000044;2' [Aug 18 10:34:31] DEBUG[15249] bridge_native_rtp.c: Bridge '724f7ab9-ed85-4748-9bb7-91218a7c6261' can not use native RTP bridge as could not get details [Aug 18 10:34:31] DEBUG[15249] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:31] DEBUG[15249] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15249] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.639': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[15249] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:732/channel:1629282871.639, detail: [Aug 18 10:34:31] DEBUG[15249] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261 is already using the new technology. [Aug 18 10:34:31] DEBUG[14895] channel.c: Channel 0x7f0c3803b8c0 'SIP/zvonobot-000000c6' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:732/channel:1629282871.639': 0x7f0c300adb10 created [Aug 18 10:34:31] DEBUG[15115] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[13852] channel.c: Channel 0x7f0c9400a450 'Recorder/ARI-00000024;2' destroying [Aug 18 10:34:31] DEBUG[13852] stasis.c: Destroying topic. name: cache:246/channel:1629282841.207, detail: [Aug 18 10:34:31] DEBUG[13852] stasis.c: Topic 'cache:246/channel:1629282841.207': 0x7f0c94063c20 destroyed [Aug 18 10:34:31] DEBUG[13852] stasis.c: Destroying topic. name: channel:1629282841.207, detail: [Aug 18 10:34:31] DEBUG[13852] stasis.c: Topic 'channel:1629282841.207': 0x7f0c94057d20 destroyed [Aug 18 10:34:31] DEBUG[15115] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:31] DEBUG[15249] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x23a5040(Recorder/ARI-00000044;2) is joining simple_bridge technology [Aug 18 10:34:31] DEBUG[15249] channel.c: Channel Recorder/ARI-00000044;2 setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[15249] channel.c: Channel SIP/zvonobot-0000003d setting write format path: slin -> ulaw [Aug 18 10:34:31] DEBUG[14844] channel.c: Channel 0x7f0c8c034ba0 'Announcer/ARI-0000003d;2' destroying [Aug 18 10:34:31] DEBUG[13777] res_rtp_asterisk.c: (0x7f0c1c0b2b20) DTLS stop [Aug 18 10:34:31] DEBUG[13777] res_rtp_asterisk.c: (0x7f0c1c0b2b20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[13777] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[13777] res_rtp_asterisk.c: (0x7f0c1c0b2b20) ICE stopped [Aug 18 10:34:31] DEBUG[13777] rtp_engine.c: Destroyed RTP instance '0x7f0c1c0b2b20' [Aug 18 10:34:31] DEBUG[13777] channel.c: Channel 0x7f0c1c120cb0 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20' destroying [Aug 18 10:34:31] DEBUG[13706] channel.c: Channel 0x7f0c9c00dcf0 'SIP/zvonobot-0000000e' destroying [Aug 18 10:34:31] DEBUG[13444] channel.c: Channel 0x7f0c180acf90 'Recorder/ARI-00000011;1' destroying [Aug 18 10:34:31] DEBUG[13447] channel.c: Channel 0x7f0c8c0178b0 'SIP/zvonobot-00000020' destroying [Aug 18 10:34:31] DEBUG[14899] channel.c: Channel 0x7f0c10106460 'Recorder/ARI-00000040;1' destroying [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212977': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212977' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:31] DEBUG[14888] bridge_channel.c: Setting 0x7f0c100718c0(Recorder/ARI-00000040;2) state from:0 to:1 [Aug 18 10:34:31] DEBUG[14947] stasis.c: Creating topic. name: channel:1629282871.640, detail: [Aug 18 10:34:31] DEBUG[15249] channel.c: Channel SIP/zvonobot-0000003d setting read format path: ulaw -> slin [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel 'robot_212977': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel 'robot_212977' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14947] stasis.c: Topic 'channel:1629282871.640': 0x7f0c700a8900 created [Aug 18 10:34:31] DEBUG[14947] stasis.c: Creating topic. name: cache:733/channel:1629282871.640, detail: [Aug 18 10:34:31] DEBUG[14947] stasis.c: Topic 'cache:733/channel:1629282871.640': 0x7f0c7000bee0 created [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:31] DEBUG[14888] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: pulling 0x7f0c100718c0(Recorder/ARI-00000040;2) [Aug 18 10:34:31] VERBOSE[14888] bridge_channel.c: Channel Recorder/ARI-00000040;2 left 'simple_bridge' stasis-bridge <79f92216-f8f4-49dd-85f1-f154853e1fd1> [Aug 18 10:34:31] DEBUG[14888] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c100718c0(Recorder/ARI-00000040;2) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:31] DEBUG[13706] stasis.c: Destroying topic. name: cache:21/channel:212977, detail: [Aug 18 10:34:31] DEBUG[13706] stasis.c: Topic 'cache:21/channel:212977': 0x7f0c9c00f9b0 destroyed [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:31] DEBUG[15248] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:31] DEBUG[13706] stasis.c: Destroying topic. name: channel:212977, detail: [Aug 18 10:34:31] DEBUG[13777] stasis.c: Destroying topic. name: cache:233/channel:robot_212977, detail: [Aug 18 10:34:31] DEBUG[13706] stasis.c: Topic 'channel:212977': 0x7f0c9c00f860 destroyed [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:31] DEBUG[13777] stasis.c: Topic 'cache:233/channel:robot_212977': 0x7f0c1c043c80 destroyed [Aug 18 10:34:31] DEBUG[14899] stasis.c: Destroying topic. name: cache:475/channel:1629282858.413, detail: [Aug 18 10:34:31] DEBUG[14899] stasis.c: Topic 'cache:475/channel:1629282858.413': 0x7f0c1004b000 destroyed [Aug 18 10:34:31] DEBUG[14899] stasis.c: Destroying topic. name: channel:1629282858.413, detail: [Aug 18 10:34:31] DEBUG[13444] stasis.c: Destroying topic. name: cache:146/channel:1629282835.122, detail: [Aug 18 10:34:31] DEBUG[14899] stasis.c: Topic 'channel:1629282858.413': 0x7f0c1004f660 destroyed [Aug 18 10:34:31] DEBUG[14844] stasis.c: Destroying topic. name: cache:524/channel:1629282860.455, detail: [Aug 18 10:34:31] DEBUG[13444] stasis.c: Topic 'cache:146/channel:1629282835.122': 0x7f0c1800c330 destroyed [Aug 18 10:34:31] DEBUG[13706] channel.c: Channel 0x7f0ca800d0a0 'Snoop/212977-0000000b' destroying [Aug 18 10:34:31] DEBUG[14844] stasis.c: Topic 'cache:524/channel:1629282860.455': 0x7f0c8c111880 destroyed [Aug 18 10:34:31] DEBUG[14844] stasis.c: Destroying topic. name: channel:1629282860.455, detail: [Aug 18 10:34:31] DEBUG[13777] stasis.c: Destroying topic. name: channel:robot_212977, detail: [Aug 18 10:34:31] DEBUG[13777] stasis.c: Topic 'channel:robot_212977': 0x7f0c1c043530 destroyed [Aug 18 10:34:31] DEBUG[13447] channel.c: Channel 0x7f0c280c6fb0 'Snoop/212995-00000007' destroying [Aug 18 10:34:31] DEBUG[13444] stasis.c: Destroying topic. name: channel:1629282835.122, detail: [Aug 18 10:34:31] DEBUG[14844] stasis.c: Topic 'channel:1629282860.455': 0x7f0c8c039210 destroyed [Aug 18 10:34:31] DEBUG[14888] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14888] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14888] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14888] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14888] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14888] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1 is already using the new technology. [Aug 18 10:34:31] DEBUG[14888] channel.c: Channel 0x7f0c1003bdc0 'Recorder/ARI-00000040;2' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[13706] stasis.c: Destroying topic. name: cache:218/channel:1629282839.183, detail: [Aug 18 10:34:31] DEBUG[13706] stasis.c: Topic 'cache:218/channel:1629282839.183': 0x7f0ca8009410 destroyed [Aug 18 10:34:31] DEBUG[13706] stasis.c: Destroying topic. name: channel:1629282839.183, detail: [Aug 18 10:34:31] DEBUG[13706] stasis.c: Topic 'channel:1629282839.183': 0x7f0ca80574d0 destroyed [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212995': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[15249] channel.c: Channel Recorder/ARI-00000044;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[15118] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212995' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[15118] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[13444] stasis.c: Topic 'channel:1629282835.122': 0x7f0c1807fc30 destroyed [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:48/channel:212995, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:48/channel:212995': 0x7f0c8c07da20 destroyed [Aug 18 10:34:31] DEBUG[15125] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30' [Aug 18 10:34:31] DEBUG[15125] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[14937] channel.c: Channel 0x7f0c1c157030 'Announcer/ARI-00000049;2' destroying [Aug 18 10:34:31] DEBUG[12971] chan_sip.c: Hangup call SIP/zvonobot-00000013, SIP callid 3ffca7933f34ff2b4725f8a61c60bd74@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[12971] res_rtp_asterisk.c: (0x7f0c1c00bc40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[12971] res_rtp_asterisk.c: (0x7f0c1c00bc40) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[15125] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15251] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15250] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:212995, detail: [Aug 18 10:34:31] DEBUG[15133] channel.c: Channel 0x7f0c10123540 'Recorder/ARI-00000055;1' allocated [Aug 18 10:34:31] DEBUG[15250] http.c: HTTP Request URI is /ari/channels/1629282839.183 [Aug 18 10:34:31] DEBUG[15133] stasis.c: Creating topic. name: channel:1629282871.641, detail: [Aug 18 10:34:31] DEBUG[15250] http.c: match request [ari/channels/1629282839.183] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15251] http.c: HTTP Request URI is /ari/channels/213011 [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:212995': 0x7f0c8c0199f0 destroyed [Aug 18 10:34:31] DEBUG[15250] http.c: match request [ari/channels/1629282839.183] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[14057] bridge_channel.c: Setting 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15250] http.c: match request [ari/channels/1629282839.183] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[14244] bridge_channel.c: Setting 0x7f0c90040640(Snoop/212982-00000010) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15250] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Finding handler for channels/1629282839.183 [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15133] stasis.c: Topic 'channel:1629282871.641': 0x7f0c10001620 created [Aug 18 10:34:31] DEBUG[15133] stasis.c: Creating topic. name: cache:734/channel:1629282871.641, detail: [Aug 18 10:34:31] DEBUG[15133] stasis.c: Topic 'cache:734/channel:1629282871.641': 0x7f0c100ea540 created [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[13447] stasis.c: Destroying topic. name: cache:149/channel:1629282835.124, detail: [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face;received=159.65.48.104 From: ;tag=as1a5706e7 To: ;tag=as6aac0e8c Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e47d332" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[13447] stasis.c: Topic 'cache:149/channel:1629282835.124': 0x7f0c280b3ac0 destroyed [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[14057] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: pulling 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] VERBOSE[14057] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 left 'simple_bridge' stasis-bridge <28c87384-44a9-4ebc-9328-4118df068e33> [Aug 18 10:34:31] DEBUG[15251] http.c: match request [ari/channels/213011] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Finding handler for 1629282839.183 [Aug 18 10:34:31] DEBUG[13447] stasis.c: Destroying topic. name: channel:1629282835.124, detail: [Aug 18 10:34:31] DEBUG[14057] bridge_channel.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33: 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking channels create: Didn't match 1629282839.183 [Aug 18 10:34:31] DEBUG[15251] http.c: match request [ari/channels/213011] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[13447] stasis.c: Topic 'channel:1629282835.124': 0x7f0c280a2df0 destroyed [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15250] res_ari.c: Checking channels externalMedia: Didn't match 1629282839.183 [Aug 18 10:34:31] DEBUG[15250] res_ari.c: No explicit handler found for 1629282839.183. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15251] http.c: match request [ari/channels/213011] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15251] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Finding handler for channels/213011 [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[14244] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: pulling 0x7f0c90040640(Snoop/212982-00000010) [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] VERBOSE[14244] bridge_channel.c: Channel Snoop/212982-00000010 left 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Finding handler for 213011 [Aug 18 10:34:31] DEBUG[14244] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c90040640(Snoop/212982-00000010) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking channels create: Didn't match 213011 [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[14057] bridge_native_rtp.c: Bridge '28c87384-44a9-4ebc-9328-4118df068e33' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14057] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15251] res_ari.c: Checking channels externalMedia: Didn't match 213011 [Aug 18 10:34:31] DEBUG[15251] res_ari.c: No explicit handler found for 213011. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[14057] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14057] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14057] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14057] bridge.c: Bridge 28c87384-44a9-4ebc-9328-4118df068e33 is already using the new technology. [Aug 18 10:34:31] DEBUG[14937] stasis.c: Destroying topic. name: cache:616/channel:1629282866.537, detail: [Aug 18 10:34:31] DEBUG[14937] stasis.c: Topic 'cache:616/channel:1629282866.537': 0x7f0c1c053830 destroyed [Aug 18 10:34:31] DEBUG[14244] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14244] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14244] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14937] stasis.c: Destroying topic. name: channel:1629282866.537, detail: [Aug 18 10:34:31] DEBUG[14244] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14244] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14244] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe is already using the new technology. [Aug 18 10:34:31] DEBUG[14937] stasis.c: Topic 'channel:1629282866.537': 0x7f0c1c0a1aa0 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:732/channel:1629282871.639, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e96face;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1a5706e7 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6aac0e8c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e47d332" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 (Checking To) --From tag as1a5706e7 --To-tag as6aac0e8c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5ec95880534c30fc1b3237ad07f2f116@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #146 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #146)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #59 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #59)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14057] bridge_channel.c: Bridge is returning 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) to write format slin16 [Aug 18 10:34:31] DEBUG[14057] channel.c: Channel UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30 setting write format path: slin16 -> slin16 [Aug 18 10:34:31] DEBUG[14244] bridge_channel.c: Bridge is returning 0x7f0c90040640(Snoop/212982-00000010) to read format slin [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:732/channel:1629282871.639': 0x7f0c300adb10 destroyed [Aug 18 10:34:31] DEBUG[14057] stasis/control.c: robot_213011, 28c87384-44a9-4ebc-9328-4118df068e33: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[14244] channel.c: Channel Snoop/212982-00000010 setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[14244] bridge_channel.c: Bridge is returning 0x7f0c90040640(Snoop/212982-00000010) to write format slin [Aug 18 10:34:31] DEBUG[14244] channel.c: Channel Snoop/212982-00000010 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[14057] stasis/app.c: bridge '28c87384-44a9-4ebc-9328-4118df068e33': is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[14011] stasis/control.c: robot_213011: Channel departing bridge [Aug 18 10:34:31] DEBUG[14011] bridge.c: Waiting for 0x7f0c280f4bc0(UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30) bridge thread to die. [Aug 18 10:34:31] DEBUG[14057] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[14244] stasis/control.c: 1629282845.251, a76fe935-dd52-4012-a523-638ab1ec4dfe: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[14011] stasis/app.c: channel 'robot_213011': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.639, detail: [Aug 18 10:34:31] DEBUG[14011] channel.c: Channel 0x7f0c240f6d50 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[14244] stasis/app.c: bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe': is 3 interested in calls_0 [Aug 18 10:34:31] DEBUG[14132] stasis/control.c: 1629282845.251: Channel departing bridge [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.639': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2;received=159.65.48.104 From: ;tag=as6400b9b5 To: ;tag=as56577292 Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ee1fa31" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK47ce3eb2;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6400b9b5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as56577292 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ee1fa31" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:31] DEBUG[14132] bridge.c: Waiting for 0x7f0c90040640(Snoop/212982-00000010) bridge thread to die. [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:31] DEBUG[14244] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[14132] stasis/app.c: channel '1629282845.251': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[14132] stasis/app.c: channel '1629282845.251' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14132] channel.c: Channel 0x7f0c88099000 'Snoop/212982-00000010' hanging up. Refs: 3 [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:18', '"" <>', '', 's', 'default', 'Snoop/212967-00000015', '', 'Stasis', 'calls_0', 10, 10, 'ANSWERED', 3, '', '1629282858.425', '')] [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 (Checking To) --From tag as6400b9b5 --To-tag as56577292 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 5b7f312012d9fa5512fdf5716b0b0558@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #160 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #160)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000c6 - start 1629282865.002721 answer 0.000000 end 1629282871.336326 dur 6.333 bill 1629282871.336 dispo NO ANSWER [Aug 18 10:34:31] DEBUG[14643] res_stasis_recording.c: 1629282859.435: Sending record(213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr.wav) command [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed;received=159.65.48.104 From: ;tag=as0f1a808c To: ;tag=as1424589c Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e89ff91" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7e3d91ed;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0f1a808c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1424589c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e89ff91" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3782ef707142714164cf352b663534ff@159.65.48.104:5060 (Checking To) --From tag as0f1a808c --To-tag as1424589c [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.642, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3782ef707142714164cf352b663534ff@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c;received=159.65.48.104 From: ;tag=as3f6b0566 To: ;tag=as2be956f9 Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dec0191" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK510b792c;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f6b0566 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2be956f9 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dec0191" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 (Checking To) --From tag as3f6b0566 --To-tag as2be956f9 [Aug 18 10:34:31] DEBUG[15252] app.c: play_and_record: , /var/spool/asterisk/recording/213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr, 'wav' [Aug 18 10:34:31] DEBUG[14643] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[14643] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15252] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:31] DEBUG[14968] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40' [Aug 18 10:34:31] VERBOSE[15252] app.c: x=0, open writing: /var/spool/asterisk/recording/213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr format: wav, 0x7f0c7c004c90 [Aug 18 10:34:31] DEBUG[14968] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[14968] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[14653] channel.c: Channel 0x7f0c3c149bd0 'Recorder/ARI-00000035;2' destroying [Aug 18 10:34:31] DEBUG[14653] stasis.c: Destroying topic. name: cache:443/channel:1629282856.385, detail: [Aug 18 10:34:31] DEBUG[14653] stasis.c: Topic 'cache:443/channel:1629282856.385': 0x7f0c3c089750 destroyed [Aug 18 10:34:31] DEBUG[14653] stasis.c: Destroying topic. name: channel:1629282856.385, detail: [Aug 18 10:34:31] DEBUG[14653] stasis.c: Topic 'channel:1629282856.385': 0x7f0c3c11c2f0 destroyed [Aug 18 10:34:31] DEBUG[15253] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15253] http.c: HTTP Request URI is /ari/channels/212995 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.642': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[15253] http.c: match request [ari/channels/212995] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[13483] bridge_channel.c: Setting 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15253] http.c: match request [ari/channels/212995] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15253] http.c: match request [ari/channels/212995] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15253] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Finding handler for channels/212995 [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[14529] res_rtp_asterisk.c: (0x2c14110) RTCP got report of 100 bytes from 178.62.121.41:15861 [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Finding handler for 212995 [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking channels create: Didn't match 212995 [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15253] res_ari.c: Checking channels externalMedia: Didn't match 212995 [Aug 18 10:34:31] DEBUG[15253] res_ari.c: No explicit handler found for 212995. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:735/channel:1629282871.642, detail: [Aug 18 10:34:31] DEBUG[13483] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: pulling 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) [Aug 18 10:34:31] VERBOSE[13483] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 left 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[13483] bridge_channel.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5: 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[13483] bridge_native_rtp.c: Bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[13483] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[13483] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13483] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13483] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[13483] bridge.c: Bridge d177377e-a80b-4ad9-826a-cece7d5abce5 is already using the new technology. [Aug 18 10:34:31] DEBUG[13483] bridge_channel.c: Bridge is returning 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) to write format slin16 [Aug 18 10:34:31] DEBUG[13483] channel.c: Channel UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40 setting write format path: slin16 -> slin16 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:735/channel:1629282871.642': 0x7f0c300adb10 created [Aug 18 10:34:31] DEBUG[13483] stasis/control.c: robot_212995, d177377e-a80b-4ad9-826a-cece7d5abce5: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13483] stasis/app.c: bridge 'd177377e-a80b-4ad9-826a-cece7d5abce5': is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[13483] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13457] stasis/control.c: robot_212995: Channel departing bridge [Aug 18 10:34:31] DEBUG[13457] bridge.c: Waiting for 0x7f0c7005a360(UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40) bridge thread to die. [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:735/channel:1629282871.642, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:735/channel:1629282871.642': 0x7f0c300adb10 destroyed [Aug 18 10:34:31] DEBUG[13457] stasis/app.c: channel 'robot_212995': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.642, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.642': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[13457] channel.c: Channel 0x7f0c3c05fae0 'UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000000e', '', 'Stasis', 'calls_0', 41, 29, 'ANSWERED', 3, '', '212977', '')] [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.643, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.643': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:736/channel:1629282871.643, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:736/channel:1629282871.643': 0x7f0c300926b0 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] WARNING[15016] app.c: No audio available on Recorder/ARI-00000046;1?? [Aug 18 10:34:31] VERBOSE[15016] app.c: User hung up [Aug 18 10:34:31] DEBUG[15016] res_stasis_recording.c: 1629282860.450: Recording complete [Aug 18 10:34:31] DEBUG[14972] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[14972] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[15150] channel.c: Channel 0x7f0c84148b70 'Announcer/ARI-0000004a;1' destroying [Aug 18 10:34:31] DEBUG[14969] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[15147] bridge_channel.c: Setting 0x7f0c84119870(Announcer/ARI-0000004a;2) state from:0 to:1 [Aug 18 10:34:31] DEBUG[14969] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15150] stasis.c: Destroying topic. name: cache:561/channel:1629282862.488, detail: [Aug 18 10:34:31] DEBUG[15150] stasis.c: Topic 'cache:561/channel:1629282862.488': 0x7f0c84036a00 destroyed [Aug 18 10:34:31] DEBUG[15016] channel.c: Channel 0x7f0c8c106b00 'Recorder/ARI-00000046;1' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[15259] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15257] http.c: HTTP opening session. Top level [Aug 18 10:34:31] ERROR[15153] channel.c: Channel Unique ID 'robot_212967' already in use by channel UnicastRTP/127.0.0.1:50044-0x7f0c7005f720(0x7f0c70121770) [Aug 18 10:34:31] DEBUG[15150] stasis.c: Destroying topic. name: channel:1629282862.488, detail: [Aug 18 10:34:31] DEBUG[15150] stasis.c: Topic 'channel:1629282862.488': 0x7f0c84149770 destroyed [Aug 18 10:34:31] DEBUG[15257] http.c: HTTP Request URI is /ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/play?media=sound%3Asilence%2F2 [Aug 18 10:34:31] DEBUG[15147] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: pulling 0x7f0c84119870(Announcer/ARI-0000004a;2) [Aug 18 10:34:31] VERBOSE[15147] bridge_channel.c: Channel Announcer/ARI-0000004a;2 left 'simple_bridge' stasis-bridge <5fd3583d-12a2-4028-9389-fce6801ffb6b> [Aug 18 10:34:31] DEBUG[15259] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_212984&app=calls_0&format=slin16&external_host=127.0.0.1%3A50017 [Aug 18 10:34:31] DEBUG[15147] bridge_channel.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b: 0x7f0c84119870(Announcer/ARI-0000004a;2) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[15257] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/play] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15153] channel.c: Channel 0x7f0c400ad640 'UnicastRTP/127.0.0.1:50044-0x7f0c400c76f0' destroying [Aug 18 10:34:31] DEBUG[15257] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/play] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:31] DEBUG[15257] http.c: match request [ari/bridges/f495d952-07a0-4425-9378-2616afbaca10/play] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15259] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15259] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15153] stasis.c: Destroying topic. name: cache:687/channel:robot_212967, detail: [Aug 18 10:34:31] DEBUG[15153] stasis.c: Topic 'cache:687/channel:robot_212967': 0x7f0c40067b90 destroyed [Aug 18 10:34:31] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) DTLS stop [Aug 18 10:34:31] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[15153] res_rtp_asterisk.c: (0x7f0c400c76f0) ICE stopped [Aug 18 10:34:31] DEBUG[15257] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15153] rtp_engine.c: Destroyed RTP instance '0x7f0c400c76f0' [Aug 18 10:34:31] DEBUG[15153] http.c: HTTP keeping session open. status_code:409 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15259] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15259] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[14974] stasis.c: Creating topic. name: channel:1629282871.644, detail: [Aug 18 10:34:31] DEBUG[14974] stasis.c: Topic 'channel:1629282871.644': 0x2c42f90 created [Aug 18 10:34:31] DEBUG[14974] stasis.c: Creating topic. name: cache:737/channel:1629282871.644, detail: [Aug 18 10:34:31] DEBUG[14821] chan_sip.c: Hangup call SIP/zvonobot-000000a6, SIP callid 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[14821] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:31] DEBUG[14821] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14821] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14821] channel.c: Channel 0x7f0cac056530 'SIP/zvonobot-000000a6' destroying [Aug 18 10:34:31] DEBUG[14974] stasis.c: Topic 'cache:737/channel:1629282871.644': 0x2c55820 created [Aug 18 10:34:31] DEBUG[15153] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Finding handler for bridges/f495d952-07a0-4425-9378-2616afbaca10/play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[14946] channel.c: Channel 0x7f0c1c0366a0 'Announcer/ARI-00000049;1' destroying [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15147] bridge_native_rtp.c: Bridge '5fd3583d-12a2-4028-9389-fce6801ffb6b' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[14946] stasis.c: Destroying topic. name: cache:553/channel:1629282862.480, detail: [Aug 18 10:34:31] DEBUG[14946] stasis.c: Topic 'cache:553/channel:1629282862.480': 0x7f0c1c126ee0 destroyed [Aug 18 10:34:31] DEBUG[14946] stasis.c: Destroying topic. name: channel:1629282862.480, detail: [Aug 18 10:34:31] DEBUG[14946] stasis.c: Topic 'channel:1629282862.480': 0x7f0c1c1326e0 destroyed [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Finding handler for f495d952-07a0-4425-9378-2616afbaca10 [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15257] res_ari.c: No explicit handler found for f495d952-07a0-4425-9378-2616afbaca10. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Finding handler for play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:31] DEBUG[15257] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:31] DEBUG[15257] stasis.c: Creating topic. name: channel:1629282871.645, detail: [Aug 18 10:34:31] DEBUG[15257] stasis.c: Topic 'channel:1629282871.645': 0x7f0c840990a0 created [Aug 18 10:34:31] DEBUG[15257] stasis.c: Creating topic. name: cache:738/channel:1629282871.645, detail: [Aug 18 10:34:31] DEBUG[15257] stasis.c: Topic 'cache:738/channel:1629282871.645': 0x7f0c84067160 created [Aug 18 10:34:31] DEBUG[15147] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15147] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15147] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15147] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15147] bridge.c: Bridge 5fd3583d-12a2-4028-9389-fce6801ffb6b is already using the new technology. [Aug 18 10:34:31] DEBUG[15147] channel.c: Channel 0x7f0c8408ee40 'Announcer/ARI-0000004a;2' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:31] DEBUG[14995] channel.c: Channel 0x7f0c180d4250 'Announcer/ARI-0000004e;2' allocated [Aug 18 10:34:31] DEBUG[14995] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:31] DEBUG[14995] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000004e;1' [Aug 18 10:34:31] DEBUG[13905] res_rtp_asterisk.c: (0x7f0c340f6d00) DTLS stop [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:31] DEBUG[15157] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0' [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15259] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:31] DEBUG[13905] res_rtp_asterisk.c: (0x7f0c340f6d00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[13905] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[13905] res_rtp_asterisk.c: (0x7f0c340f6d00) ICE stopped [Aug 18 10:34:31] DEBUG[13905] rtp_engine.c: Destroyed RTP instance '0x7f0c340f6d00' [Aug 18 10:34:31] DEBUG[13905] channel.c: Channel 0x7f0c34028b90 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00' destroying [Aug 18 10:34:31] DEBUG[15259] netsock2.c: Splitting '127.0.0.1:50017' into... [Aug 18 10:34:31] DEBUG[15260] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c1808f6e0(Announcer/ARI-0000004e;2) is joining [Aug 18 10:34:31] DEBUG[14991] channel.c: Channel 0x7f0c1c06a350 'Announcer/ARI-0000004f;2' allocated [Aug 18 10:34:31] DEBUG[14991] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:31] DEBUG[14991] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000004f;1' [Aug 18 10:34:31] DEBUG[15259] netsock2.c: ...host '127.0.0.1' and port '50017'. [Aug 18 10:34:31] DEBUG[15259] netsock2.c: Splitting '127.0.0.1:50017' into... [Aug 18 10:34:31] DEBUG[14756] stasis/app.c: channel 'robot_212991': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[14756] channel.c: Channel 0x7f0c38071710 'UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[15259] netsock2.c: ...host '127.0.0.1' and port '50017'. [Aug 18 10:34:31] DEBUG[15259] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:31] DEBUG[15259] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c80021ed0' [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) RTP allocated port 18454 [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) ICE creating session 127.0.0.1:18454 (18454) [Aug 18 10:34:31] DEBUG[14828] channel.c: Channel 0x7f0c3800b750 'Snoop/213038-0000001b' allocated [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) ICE create [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) ICE add system candidates [Aug 18 10:34:31] DEBUG[15259] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:31] DEBUG[15259] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) ICE add candidate: 159.65.48.104:18454, 2130706431 [Aug 18 10:34:31] DEBUG[15259] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:31] DEBUG[15259] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:31] DEBUG[15259] res_rtp_asterisk.c: (0x7f0c80021ed0) ICE add candidate: 10.131.0.10:18454, 2130706431 [Aug 18 10:34:31] DEBUG[15259] rtp_engine.c: RTP instance '0x7f0c80021ed0' is setup and ready to go [Aug 18 10:34:31] DEBUG[15259] stasis.c: Creating topic. name: channel:robot_212984, detail: [Aug 18 10:34:31] DEBUG[15157] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[15157] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:31] DEBUG[14828] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[14828] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[14519] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d'. Checking compatability for channels 'SIP/zvonobot-00000049' and 'Recorder/ARI-0000003b;2' [Aug 18 10:34:31] DEBUG[14519] bridge_native_rtp.c: Bridge '26acc09b-99c1-4bbb-afbd-344c8a9a505d' can not use native RTP bridge as channel 'SIP/zvonobot-00000049' has features which prevent it [Aug 18 10:34:31] DEBUG[14519] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:31] DEBUG[14519] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14519] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15259] stasis.c: Topic 'channel:robot_212984': 0x7f0c80012600 created [Aug 18 10:34:31] DEBUG[14519] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14519] bridge.c: Bridge 26acc09b-99c1-4bbb-afbd-344c8a9a505d is already using the new technology. [Aug 18 10:34:31] DEBUG[15262] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15259] stasis.c: Creating topic. name: cache:739/channel:robot_212984, detail: [Aug 18 10:34:31] DEBUG[15260] bridge_channel.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: pushing 0x7f0c1808f6e0(Announcer/ARI-0000004e;2) [Aug 18 10:34:31] DEBUG[15267] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213131': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[15261] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c1c126ff0(Announcer/ARI-0000004f;2) is joining [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776;received=159.65.48.104 From: ;tag=as1180a433 To: ;tag=as43ab9e8c Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="28a56691" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[15259] stasis.c: Topic 'cache:739/channel:robot_212984': 0x7f0c80046df0 created [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213131' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[15267] http.c: HTTP Request URI is /ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/play?media=sound%3Asilence%2F2 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39bc6776;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1180a433 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as43ab9e8c [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="28a56691" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[15260] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:736/channel:1629282871.643, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:736/channel:1629282871.643': 0x7f0c300926b0 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.643, detail: [Aug 18 10:34:31] VERBOSE[15260] bridge_channel.c: Channel Announcer/ARI-0000004e;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[15267] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/play] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15262] http.c: HTTP Request URI is /ari/channels/212991 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 (Checking To) --From tag as1180a433 --To-tag as43ab9e8c [Aug 18 10:34:31] DEBUG[15267] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/play] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15261] bridge_channel.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: pushing 0x7f0c1c126ff0(Announcer/ARI-0000004f;2) [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:416/channel:213131, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 467c381324b5f4ee7e9517717267c0e6@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:416/channel:213131': 0x7f0cac049550 destroyed [Aug 18 10:34:31] DEBUG[15262] http.c: match request [ari/channels/212991] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[15269] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:213131, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[15261] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:31] VERBOSE[15261] bridge_channel.c: Channel Announcer/ARI-0000004f;2 joined 'simple_bridge' stasis-bridge <94ef4fc9-246b-4999-9567-b41f8ba44681> [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:213131': 0x7f0cac0740b0 destroyed [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15260] bridge.c: Chose bridge technology softmix [Aug 18 10:34:31] VERBOSE[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: switching from simple_bridge technology to softmix [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling softmix technology constructor [Aug 18 10:34:31] DEBUG[15262] http.c: match request [ari/channels/212991] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: moving 0x7f0c40072e10(SIP/zvonobot-00000029) to dummy bridge temporarily [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: moving 0x7f0c40070f90(Recorder/ARI-00000036;2) to dummy bridge temporarily [Aug 18 10:34:31] DEBUG[15261] bridge_native_rtp.c: Bridge '94ef4fc9-246b-4999-9567-b41f8ba44681' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40072e10(SIP/zvonobot-00000029) is leaving simple_bridge technology (dummy) [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40070f90(Recorder/ARI-00000036;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:31] DEBUG[15267] http.c: match request [ari/bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/play] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15261] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling simple_bridge technology stop [Aug 18 10:34:31] DEBUG[15262] http.c: match request [ari/channels/212991] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15262] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Finding handler for channels/212991 [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Finding handler for 212991 [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking channels create: Didn't match 212991 [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15262] res_ari.c: Checking channels externalMedia: Didn't match 212991 [Aug 18 10:34:31] DEBUG[15262] res_ari.c: No explicit handler found for 212991. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c1808f6e0(Announcer/ARI-0000004e;2) is joining softmix technology [Aug 18 10:34:31] DEBUG[15168] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[15168] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15261] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15267] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.643': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Announcer/ARI-0000004e;2: [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:00', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212977', '')] [Aug 18 10:34:31] DEBUG[15260] channel.c: Channel Announcer/ARI-0000004e;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[14519] audiohook.c: Audiohook 0x7f0c38120330 has stale audio in its factories. Flushing them both [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel 'robot_213007': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel 'robot_213007' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:262/channel:robot_213007, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:262/channel:robot_213007': 0x7f0c3402ad50 destroyed [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213007, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:robot_213007': 0x7f0c340fce20 destroyed [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.647, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.647': 0x7f0c300a9260 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:740/channel:1629282871.647, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:740/channel:1629282871.647': 0x7f0c300413e0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:740/channel:1629282871.647, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:740/channel:1629282871.647': 0x7f0c300413e0 destroyed [Aug 18 10:34:31] DEBUG[15261] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15261] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Announcer/ARI-0000004e;2: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Announcer/ARI-0000004e;2: Not in SFU mode [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40072e10(SIP/zvonobot-00000029) is joining softmix technology [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: SIP/zvonobot-00000029: [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: SIP/zvonobot-00000029: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: SIP/zvonobot-00000029: Not in SFU mode [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: 0x7f0c40070f90(Recorder/ARI-00000036;2) is joining softmix technology [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Recorder/ARI-00000036;2: [Aug 18 10:34:31] DEBUG[15260] channel.c: Channel Recorder/ARI-00000036;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:31] DEBUG[15260] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Recorder/ARI-00000036;2: [Aug 18 10:34:31] DEBUG[15260] bridge_softmix.c: Recorder/ARI-00000036;2: Not in SFU mode [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling softmix technology start [Aug 18 10:34:31] DEBUG[15260] bridge.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: calling simple_bridge technology destructor [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:31] DEBUG[15269] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213038&app=calls_0&format=slin16&external_host=127.0.0.1%3A50035 [Aug 18 10:34:31] DEBUG[15261] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681 is already using the new technology. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.647, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.647': 0x7f0c300a9260 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:48', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000020', '', 'Stasis', 'calls_0', 38, 31, 'ANSWERED', 3, '', '212995', '')] [Aug 18 10:34:31] DEBUG[15261] bridge.c: Bridge 94ef4fc9-246b-4999-9567-b41f8ba44681: 0x7f0c1c126ff0(Announcer/ARI-0000004f;2) is joining simple_bridge technology [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15271] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15269] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Finding handler for bridges/26acc09b-99c1-4bbb-afbd-344c8a9a505d/play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.648, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.648': 0x7f0c300413e0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:741/channel:1629282871.648, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:741/channel:1629282871.648': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:741/channel:1629282871.648, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:741/channel:1629282871.648': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.648, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.648': 0x7f0c300413e0 destroyed [Aug 18 10:34:31] DEBUG[15271] http.c: HTTP Request URI is /ari/channels/1629282842.212 [Aug 18 10:34:31] DEBUG[15269] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15263] stasis/app.c: Channel '1629282869.603' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[15263] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 80 instead [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #33 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15270] bridge_softmix.c: Bridge fa4500bd-5aa0-4ef0-bd71-2993d47cd759: starting mixing thread [Aug 18 10:34:31] DEBUG[14991] res_stasis_playback.c: 1629282867.555: Sending play(sound:silence/2) command [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #33)) [Aug 18 10:34:31] DEBUG[14991] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[14995] res_stasis_playback.c: 1629282867.554: Sending play(sound:silence/2) command [Aug 18 10:34:31] DEBUG[15271] http.c: match request [ari/channels/1629282842.212] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[14991] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15008] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[15008] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[14995] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:59', '"" <>', '', 's', 'default', 'Snoop/212977-0000000b', 'UnicastRTP/127.0.0.1:50194-0x7f0c1c0b2b20', 'Stasis', 'calls_0', 29, 29, 'ANSWERED', 3, '', '1629282839.183', '')] [Aug 18 10:34:31] DEBUG[14995] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[14827] chan_sip.c: Hangup call SIP/zvonobot-000000be, SIP callid 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[14827] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:31] DEBUG[14827] res_rtp_asterisk.c: (0x7f0c18009d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14827] res_rtp_asterisk.c: (0x7f0c18009d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14827] channel.c: Channel 0x7f0c18029cf0 'SIP/zvonobot-000000be' destroying [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[14394] audiohook.c: Audiohook 0x7f0c8c14b960 has stale audio in its factories. Flushing them both [Aug 18 10:34:31] DEBUG[14394] res_rtp_asterisk.c: (0x7f0c38023850) RTP ooh, format changed from none to ulaw [Aug 18 10:34:31] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:31] DEBUG[15269] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15271] http.c: match request [ari/channels/1629282842.212] with handler [phoneprov] len 9 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116861@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7facccd0 Max-Forwards: 70 From: ;tag=as29706635 To: Contact: Call-ID: 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 997001768 997001768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18760 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (2) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '47a6c48d7b325c224dcbc6011b5d56ed@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cac01e130) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cac01e130' [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872;received=159.65.48.104 From: ;tag=as7d998899 To: ;tag=as1d0a14f1 Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31980f65" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69434872;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d998899 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1d0a14f1 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31980f65" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 (Checking To) --From tag as7d998899 --To-tag as1d0a14f1 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #150 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #150)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15271] http.c: match request [ari/channels/1629282842.212] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:31] DEBUG[15269] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15271] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #23 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Finding handler for 26acc09b-99c1-4bbb-afbd-344c8a9a505d [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Finding handler for channels/1629282842.212 [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Finding handler for 1629282842.212 [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking channels create: Didn't match 1629282842.212 [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15271] res_ari.c: Checking channels externalMedia: Didn't match 1629282842.212 [Aug 18 10:34:31] DEBUG[15271] res_ari.c: No explicit handler found for 1629282842.212. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15267] res_ari.c: No explicit handler found for 26acc09b-99c1-4bbb-afbd-344c8a9a505d. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Finding handler for play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:31] DEBUG[15267] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:31] DEBUG[15267] stasis.c: Creating topic. name: channel:1629282871.649, detail: [Aug 18 10:34:31] DEBUG[15267] stasis.c: Topic 'channel:1629282871.649': 0x7f0ca8009710 created [Aug 18 10:34:31] DEBUG[15267] stasis.c: Creating topic. name: cache:742/channel:1629282871.649, detail: [Aug 18 10:34:31] DEBUG[15267] stasis.c: Topic 'cache:742/channel:1629282871.649': 0x7f0ca802d890 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #23)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #125 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #125)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1ef0f4b2474f40f224b8353e16bf4d21@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18009d30) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18009d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18009d30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c18009d30) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c18009d30' [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK39a22c32 Max-Forwards: 70 From: ;tag=as70b1d74e To: ;tag=as39a2ec19 Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1351bf1c", response="b8611a412263fa5a6c170456fdc9a68e" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK39a22c32 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213154': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '213154' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:516/channel:213154, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as70b1d74e [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:516/channel:213154': 0x7f0c1802c4f0 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as39a2ec19 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:213154, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:213154': 0x7f0c1802ba70 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.650, detail: [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15269] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="1351bf1c", response="b8611a412263fa5a6c170456fdc9a68e" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 (Checking From) --From tag as70b1d74e --To-tag as39a2ec19 [Aug 18 10:34:31] DEBUG[15269] netsock2.c: Splitting '127.0.0.1:50035' into... [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.650': 0x7f0c300e94b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:743/channel:1629282871.650, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:743/channel:1629282871.650': 0x7f0c300a9260 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:743/channel:1629282871.650, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:743/channel:1629282871.650': 0x7f0c300a9260 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.650, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.650': 0x7f0c300e94b0 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:55', '"" <>', '', 's', 'default', 'Snoop/212995-00000007', 'UnicastRTP/127.0.0.1:50220-0x7f0c3c052f40', 'Stasis', 'calls_0', 33, 33, 'ANSWERED', 3, '', '1629282835.124', '')] [Aug 18 10:34:31] DEBUG[15273] channel.c: Channel Announcer/ARI-0000004e;1 setting write format path: gsm -> slin [Aug 18 10:34:31] DEBUG[15269] netsock2.c: ...host '127.0.0.1' and port '50035'. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:31] DEBUG[20545] cdr.c: Finalized CDR for Snoop/212982-00000010 - start 1629282847.466188 answer 1629282847.466188 end 1629282871.407249 dur 23.941 bill 23.941 dispo ANSWERED [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15272] channel.c: Channel Announcer/ARI-0000004f;1 setting write format path: gsm -> slin [Aug 18 10:34:31] DEBUG[15269] netsock2.c: Splitting '127.0.0.1:50035' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:31] DEBUG[15185] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[15185] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15272] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15274] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c34009e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK39a22c32;received=178.62.121.41 From: ;tag=as70b1d74e To: ;tag=as39a2ec19 Call-ID: 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #159 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #159)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Session timer stopped: 118 - 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15269] netsock2.c: ...host '127.0.0.1' and port '50035'. [Aug 18 10:34:31] DEBUG[15274] http.c: HTTP Request URI is /ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:34:31] DEBUG[15274] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15274] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15274] http.c: match request [ari/bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6] with handler [ari] len 3 [Aug 18 10:34:31] VERBOSE[15272] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 From: ;tag=as563f7715 To: ;tag=as705131ac Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as563f7715 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as705131ac [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c5c3587" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 (Checking To) --From tag as563f7715 --To-tag as705131ac [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '30cb159f67289df002568fe9006f4752@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116902@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK327746ca Max-Forwards: 70 From: ;tag=as563f7715 To: Contact: Call-ID: 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #152 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #152)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd;received=159.65.48.104 From: ;tag=as3f0bc324 To: ;tag=as797a96ff Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5358a1b7" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK16e63ebd;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3f0bc324 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as797a96ff [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5358a1b7" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 (Checking To) --From tag as3f0bc324 --To-tag as797a96ff [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 71c549ac1adedf1f733725e63c013547@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae;received=159.65.48.104 From: ;tag=as406d4539 To: ;tag=as27a3d6f4 Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="222035a3" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5942bfae;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as406d4539 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as27a3d6f4 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="222035a3" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 (Checking To) --From tag as406d4539 --To-tag as27a3d6f4 [Aug 18 10:34:31] DEBUG[15274] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Finding handler for bridges/378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[15273] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:31] VERBOSE[15273] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14295] bridge_channel.c: Setting 0x7f0c300a0130(SIP/zvonobot-00000006) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[14295] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: pulling 0x7f0c300a0130(SIP/zvonobot-00000006) [Aug 18 10:34:31] VERBOSE[14295] bridge_channel.c: Channel SIP/zvonobot-00000006 left 'simple_bridge' stasis-bridge [Aug 18 10:34:31] DEBUG[14295] bridge_channel.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d: 0x7f0c300a0130(SIP/zvonobot-00000006) is leaving simple_bridge technology [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 From: ;tag=as3056f2e0 To: ;tag=as20b331c1 Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK15b19ac1;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3056f2e0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as20b331c1 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2529fca1" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 (Checking To) --From tag as3056f2e0 --To-tag as20b331c1 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 6cd1c9a6103a0e4a04e8762f085cdf7f@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15269] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:31] DEBUG[15269] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c0696e0' [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) RTP allocated port 11786 [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) ICE creating session 127.0.0.1:11786 (11786) [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) ICE create [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Finding handler for 378d72c1-dd9d-472b-9f36-5c575a6102e6 [Aug 18 10:34:31] DEBUG[15274] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15274] res_ari.c: No explicit handler found for 378d72c1-dd9d-472b-9f36-5c575a6102e6. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[15274] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: telling all channels to leave the party [Aug 18 10:34:31] DEBUG[15274] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:31] DEBUG[15274] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: queueing action type:13 sub:1001 [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:31] DEBUG[15274] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[14295] bridge_native_rtp.c: Bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling stasis bridge destructor [Aug 18 10:34:31] DEBUG[15010] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: bridge '378d72c1-dd9d-472b-9f36-5c575a6102e6' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14295] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15010] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.651, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.651': 0x7f0c300adb10 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:744/channel:1629282871.651, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:744/channel:1629282871.651': 0x7f0c30065c30 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:744/channel:1629282871.651, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:744/channel:1629282871.651': 0x7f0c30065c30 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.651, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.651': 0x7f0c300adb10 destroyed [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology stop [Aug 18 10:34:31] DEBUG[14295] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15003] channel.c: Channel 0x7f0c30065e10 'Recorder/ARI-00000050;2' allocated [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 378d72c1-dd9d-472b-9f36-5c575a6102e6: calling simple_bridge technology destructor [Aug 18 10:34:31] DEBUG[15274] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15003] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 From: ;tag=as4d13c830 To: ;tag=as7af717bc Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[14295] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15193] channel.c: Channel 0x7f0c181092d0 'Announcer/ARI-00000056;1' allocated [Aug 18 10:34:31] DEBUG[14295] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15193] stasis.c: Creating topic. name: channel:1629282871.652, detail: [Aug 18 10:34:31] DEBUG[15193] stasis.c: Topic 'channel:1629282871.652': 0x7f0c1807c900 created [Aug 18 10:34:31] DEBUG[15193] stasis.c: Creating topic. name: cache:745/channel:1629282871.652, detail: [Aug 18 10:34:31] DEBUG[15193] stasis.c: Topic 'cache:745/channel:1629282871.652': 0x7f0c1800aae0 created [Aug 18 10:34:31] DEBUG[14295] bridge.c: Bridge fa1a4da9-c446-4fa8-95aa-bada67702e1d is already using the new technology. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[15276] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3185f809;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) ICE add system candidates [Aug 18 10:34:31] DEBUG[20534] stasis.c: Destroying topic. name: cache:68/bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6, detail: [Aug 18 10:34:31] DEBUG[20534] stasis.c: Topic 'cache:68/bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6': 0x7f0cb006aa70 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:23', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000a6', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213131', '')] [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4d13c830 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7af717bc [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20534] stasis.c: Destroying topic. name: bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6, detail: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[15276] http.c: HTTP Request URI is /ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:34:31] DEBUG[14295] bridge_channel.c: Bridge is returning 0x7f0c300a0130(SIP/zvonobot-00000006) to read format alaw [Aug 18 10:34:31] DEBUG[20534] stasis.c: Topic 'bridge:378d72c1-dd9d-472b-9f36-5c575a6102e6': 0x7f0cb0036d50 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.653, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.653': 0x7f0c30019a70 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:746/channel:1629282871.653, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:746/channel:1629282871.653': 0x7f0c300b9d40 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:746/channel:1629282871.653, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:746/channel:1629282871.653': 0x7f0c300b9d40 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.653, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.653': 0x7f0c30019a70 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50353-0x7f0c340f6d00', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213007', '')] [Aug 18 10:34:31] DEBUG[15277] bridge_channel.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: 0x7f0c300d5740(Recorder/ARI-00000050;2) is joining [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[14295] channel.c: Channel SIP/zvonobot-00000006 setting read format path: ulaw -> alaw [Aug 18 10:34:31] DEBUG[14295] bridge_channel.c: Bridge is returning 0x7f0c300a0130(SIP/zvonobot-00000006) to write format alaw [Aug 18 10:34:31] DEBUG[14295] channel.c: Channel SIP/zvonobot-00000006 setting write format path: alaw -> ulaw [Aug 18 10:34:31] DEBUG[14295] stasis/control.c: 212969, fa1a4da9-c446-4fa8-95aa-bada67702e1d: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[14295] stasis/app.c: bridge 'fa1a4da9-c446-4fa8-95aa-bada67702e1d': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[12894] stasis/control.c: 212969: Channel departing bridge [Aug 18 10:34:31] DEBUG[12894] bridge.c: Waiting for 0x7f0c300a0130(SIP/zvonobot-00000006) bridge thread to die. [Aug 18 10:34:31] DEBUG[15276] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15195] channel.c: Channel 0x7f0c24144730 'UnicastRTP/127.0.0.1:50315-0x7f0c240f1de0' allocated [Aug 18 10:34:31] DEBUG[14295] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[15195] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:31] VERBOSE[15195] res_rtp_asterisk.c: 0x7f0c24058e90 -- Strict RTP learning after remote address set to: 127.0.0.1:50315 [Aug 18 10:34:31] DEBUG[15195] res_stasis.c: calls_0: Subscribing to robot_213015 [Aug 18 10:34:31] DEBUG[15195] stasis/app.c: Channel 'robot_213015' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[12894] stasis/app.c: channel '212969': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[12894] channel.c: Channel 0x7f0c340114f0 'SIP/zvonobot-00000006' hanging up. Refs: 3 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56bdd0ce" [Aug 18 10:34:31] DEBUG[15276] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 (Checking To) --From tag as4d13c830 --To-tag as7af717bc [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 14ff7e1762e5b2fa21c7bf7c114ce7b5@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce;received=159.65.48.104 From: ;tag=as220fc8e1 To: ;tag=as2d306c83 Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f8b6b29" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as220fc8e1 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2d306c83 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f8b6b29" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 (Checking To) --From tag as220fc8e1 --To-tag as2d306c83 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:31] DEBUG[15276] http.c: match request [ari/bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Aug 18 10:34:31] DEBUG[15269] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:31] DEBUG[15269] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) ICE add candidate: 159.65.48.104:11786, 2130706431 [Aug 18 10:34:31] DEBUG[15269] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:31] DEBUG[15269] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:31] DEBUG[15269] res_rtp_asterisk.c: (0x7f0c9c0696e0) ICE add candidate: 10.131.0.10:11786, 2130706431 [Aug 18 10:34:31] DEBUG[15269] rtp_engine.c: RTP instance '0x7f0c9c0696e0' is setup and ready to go [Aug 18 10:34:31] DEBUG[15269] stasis.c: Creating topic. name: channel:robot_213038, detail: [Aug 18 10:34:31] DEBUG[15269] stasis.c: Topic 'channel:robot_213038': 0x7f0c9c00fbe0 created [Aug 18 10:34:31] DEBUG[15269] stasis.c: Creating topic. name: cache:747/channel:robot_213038, detail: [Aug 18 10:34:31] DEBUG[15269] stasis.c: Topic 'cache:747/channel:robot_213038': 0x7f0c9c09d750 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:31] DEBUG[15195] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15195] http.c: HTTP closing session. Top level [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0397f8ce Max-Forwards: 70 From: ;tag=as220fc8e1 To: ;tag=as2d306c83 Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:31] DEBUG[15276] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Finding handler for bridges/3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Finding handler for 3f704757-87e2-45e5-8aa9-92ed6ea9feee [Aug 18 10:34:31] DEBUG[15276] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15276] res_ari.c: No explicit handler found for 3f704757-87e2-45e5-8aa9-92ed6ea9feee. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[15276] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: telling all channels to leave the party [Aug 18 10:34:31] DEBUG[15276] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:31] DEBUG[15276] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: queueing action type:13 sub:1001 [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:31] DEBUG[15276] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: calling stasis bridge destructor [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: calling simple_bridge technology stop [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 3f704757-87e2-45e5-8aa9-92ed6ea9feee: calling simple_bridge technology destructor [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: bridge '3f704757-87e2-45e5-8aa9-92ed6ea9feee' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:72/bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:72/bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee': 0x7f0c18012b90 destroyed [Aug 18 10:34:31] DEBUG[14673] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'bridge:3f704757-87e2-45e5-8aa9-92ed6ea9feee': 0x7f0c1800caa0 destroyed [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Audio is at 16258 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK437e89d6 Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116864@178.62.121.41", nonce="3f8b6b29", response="43c73b210726c419b446d559637382e6" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266557 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15277] bridge_channel.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: pushing 0x7f0c300d5740(Recorder/ARI-00000050;2) [Aug 18 10:34:31] DEBUG[15276] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.655, detail: [Aug 18 10:34:31] DEBUG[13346] channel.c: Channel 0x7f0ca003f150 'Announcer/ARI-0000000b;1' destroying [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.655': 0x7f0c30116380 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:748/channel:1629282871.655, detail: [Aug 18 10:34:31] DEBUG[15018] channel.c: Channel 0x7f0c9c0aece0 'Recorder/ARI-00000051;2' allocated [Aug 18 10:34:31] DEBUG[15018] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[15184] channel.c: Channel 0x2c6c800 'SIP/zvonobot-000000f0' allocated [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] DEBUG[15277] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:748/channel:1629282871.655': 0x7f0c300af020 created [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[15184] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[15279] bridge_channel.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: 0x7f0c9c0271a0(Recorder/ARI-00000051;2) is joining [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:748/channel:1629282871.655, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:748/channel:1629282871.655': 0x7f0c300af020 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.655, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.655': 0x7f0c30116380 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:23', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000be', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213154', '')] [Aug 18 10:34:31] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:31] DEBUG[13346] stasis.c: Destroying topic. name: cache:113/channel:1629282833.94, detail: [Aug 18 10:34:31] DEBUG[13346] stasis.c: Topic 'cache:113/channel:1629282833.94': 0x7f0ca0041060 destroyed [Aug 18 10:34:31] DEBUG[13346] stasis.c: Destroying topic. name: channel:1629282833.94, detail: [Aug 18 10:34:31] DEBUG[13346] stasis.c: Topic 'channel:1629282833.94': 0x7f0ca0035320 destroyed [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36;received=159.65.48.104 From: ;tag=as71bf7e35 To: ;tag=as3228a659 Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35c6cd6f" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as71bf7e35 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3228a659 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35c6cd6f" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] VERBOSE[15277] bridge_channel.c: Channel Recorder/ARI-00000050;2 joined 'simple_bridge' stasis-bridge <3e9f0826-47ef-4258-a05a-53af5ce8577c> [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 (Checking To) --From tag as71bf7e35 --To-tag as3228a659 [Aug 18 10:34:31] DEBUG[15184] res_stasis.c: calls_0: Subscribing to 213204 [Aug 18 10:34:31] DEBUG[15184] stasis/app.c: Channel '213204' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000006 - start 1629282822.188078 answer 1629282850.650980 end 1629282871.637823 dur 49.449 bill 20.986 dispo ANSWERED [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116880@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK117b0f36 Max-Forwards: 70 From: ;tag=as71bf7e35 To: Contact: Call-ID: 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Outgoing Call for 79821116836 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:31] VERBOSE[15278] dial.c: Called 127.0.0.1:50315 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15184] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15184] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:31] VERBOSE[15280] chan_sip.c: Audio is at 12210 [Aug 18 10:34:31] VERBOSE[15280] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:31] DEBUG[15041] stasis.c: Creating topic. name: channel:1629282871.656, detail: [Aug 18 10:34:31] DEBUG[15041] stasis.c: Topic 'channel:1629282871.656': 0x7f0c10077530 created [Aug 18 10:34:31] DEBUG[15041] stasis.c: Creating topic. name: cache:749/channel:1629282871.656, detail: [Aug 18 10:34:31] DEBUG[15041] stasis.c: Topic 'cache:749/channel:1629282871.656': 0x7f0c1004c190 created [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7;received=159.65.48.104 From: ;tag=as06a66f45 To: ;tag=as15b999c2 Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="749a75da" Content-Length: 0 <-------------> [Aug 18 10:34:31] VERBOSE[15280] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:31] DEBUG[15041] channel.c: Channel 0x7f0c100436a0 'Snoop/213027-0000001d' allocated [Aug 18 10:34:31] DEBUG[15277] bridge_native_rtp.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c'. Checking compatability for channels 'SIP/zvonobot-0000003e' and 'Recorder/ARI-00000050;2' [Aug 18 10:34:31] DEBUG[15277] bridge_native_rtp.c: Bridge '3e9f0826-47ef-4258-a05a-53af5ce8577c' can not use native RTP bridge as could not get details [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] VERBOSE[15278] dial.c: UnicastRTP/127.0.0.1:50315-0x7f0c240f1de0 answered [Aug 18 10:34:31] DEBUG[14683] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb'. Checking compatability for channels 'SIP/zvonobot-0000003c' and 'Recorder/ARI-00000046;2' [Aug 18 10:34:31] DEBUG[14683] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb' can not use native RTP bridge as channel 'SIP/zvonobot-0000003c' has features which prevent it [Aug 18 10:34:31] DEBUG[14683] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14683] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14683] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14683] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14683] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb is already using the new technology. [Aug 18 10:34:31] DEBUG[15277] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c2a1dd7;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[15041] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] VERBOSE[15280] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:31] DEBUG[15041] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Initializing initreq for method INVITE - callid 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116836@178.62.121.41 SIP/2.0 [Aug 18 10:34:31] DEBUG[15279] bridge_channel.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: pushing 0x7f0c9c0271a0(Recorder/ARI-00000051;2) [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 3 [ 52]: From: ;tag=as2d4671b8 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 6 [ 60]: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:31 GMT [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] VERBOSE[15280] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116836@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 Max-Forwards: 70 From: ;tag=as2d4671b8 To: Contact: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1289209748 1289209748 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #127 [Aug 18 10:34:31] DEBUG[15280] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[15280] dial.c: Called zvonobot/79821116836 [Aug 18 10:34:31] DEBUG[15284] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15284] http.c: HTTP Request URI is /ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/play?media=sound%3Asilence%2F2 [Aug 18 10:34:31] VERBOSE[15278] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50315-0x7f0c240f1de0 [Aug 18 10:34:31] DEBUG[15279] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:31] DEBUG[15287] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as06a66f45 [Aug 18 10:34:31] VERBOSE[15279] bridge_channel.c: Channel Recorder/ARI-00000051;2 joined 'simple_bridge' stasis-bridge <0b66d66e-5f5c-4963-a022-79b61565f473> [Aug 18 10:34:31] DEBUG[15277] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15284] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/play] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:31] DEBUG[15287] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213027&app=calls_0&format=slin16&external_host=127.0.0.1%3A50267 [Aug 18 10:34:31] DEBUG[15277] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15277] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15277] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c is already using the new technology. [Aug 18 10:34:31] DEBUG[15277] bridge.c: Bridge 3e9f0826-47ef-4258-a05a-53af5ce8577c: 0x7f0c300d5740(Recorder/ARI-00000050;2) is joining simple_bridge technology [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel Recorder/ARI-00000050;2 setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel SIP/zvonobot-0000003e setting write format path: slin -> ulaw [Aug 18 10:34:31] DEBUG[15284] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/play] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15278] stasis/app.c: Channel 'robot_213015' is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel SIP/zvonobot-0000003e setting read format path: ulaw -> slin [Aug 18 10:34:31] DEBUG[15287] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15284] http.c: match request [ari/bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/play] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15284] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Finding handler for bridges/6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb/play [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as15b999c2 [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Finding handler for 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb [Aug 18 10:34:31] DEBUG[15287] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel Recorder/ARI-00000050;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:31] DEBUG[15284] res_ari.c: No explicit handler found for 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15279] bridge_native_rtp.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473'. Checking compatability for channels 'SIP/zvonobot-0000002d' and 'Recorder/ARI-00000051;2' [Aug 18 10:34:31] DEBUG[15279] bridge_native_rtp.c: Bridge '0b66d66e-5f5c-4963-a022-79b61565f473' can not use native RTP bridge as could not get details [Aug 18 10:34:31] DEBUG[15279] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15279] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15279] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15279] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15279] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473 is already using the new technology. [Aug 18 10:34:31] DEBUG[15279] bridge.c: Bridge 0b66d66e-5f5c-4963-a022-79b61565f473: 0x7f0c9c0271a0(Recorder/ARI-00000051;2) is joining simple_bridge technology [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel Recorder/ARI-00000051;2 setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel SIP/zvonobot-0000002d setting write format path: slin -> ulaw [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel SIP/zvonobot-0000002d setting read format path: ulaw -> slin [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel Recorder/ARI-00000051;2 setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[15018] res_stasis_recording.c: 1629282867.567: Sending record(213010_RLEHAKgKjztapoIEIJWLtEnzYYryiZuH.wav) command [Aug 18 10:34:31] DEBUG[15287] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15287] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="749a75da" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 (Checking To) --From tag as06a66f45 --To-tag as15b999c2 [Aug 18 10:34:31] DEBUG[15018] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[15018] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Finding handler for play [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:31] DEBUG[15040] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0' [Aug 18 10:34:31] DEBUG[15040] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15040] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15288] app.c: play_and_record: , /var/spool/asterisk/recording/213010_RLEHAKgKjztapoIEIJWLtEnzYYryiZuH, 'wav' [Aug 18 10:34:31] DEBUG[15288] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:31] VERBOSE[15288] app.c: x=0, open writing: /var/spool/asterisk/recording/213010_RLEHAKgKjztapoIEIJWLtEnzYYryiZuH format: wav, 0x7f0c2c0930a0 [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15281] stasis/app.c: Channel '1629282871.656' is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[15281] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15290] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '062e901479878f3469dc381c0f75eb83@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:31] DEBUG[15289] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15287] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:31] DEBUG[15287] netsock2.c: Splitting '127.0.0.1:50267' into... [Aug 18 10:34:31] DEBUG[15287] netsock2.c: ...host '127.0.0.1' and port '50267'. [Aug 18 10:34:31] DEBUG[15287] netsock2.c: Splitting '127.0.0.1:50267' into... [Aug 18 10:34:31] DEBUG[15287] netsock2.c: ...host '127.0.0.1' and port '50267'. [Aug 18 10:34:31] DEBUG[15287] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:31] DEBUG[15287] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c2001ddd0' [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[13379] bridge_channel.c: Setting 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15290] http.c: HTTP Request URI is /ari/channels/212981 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15003] res_stasis_recording.c: 1629282867.563: Sending record(213025_lNYdPoaZyNgkYDodNHVSYmzebqmBqduB.wav) command [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2613fd3872cf0f7c728c04a525975962@159.65.48.104:5060' Method: BYE [Aug 18 10:34:31] DEBUG[15289] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:31] DEBUG[15019] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80' [Aug 18 10:34:31] DEBUG[15019] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel Recorder/ARI-00000051;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:31] DEBUG[15020] channel.c: Soft-Hanging (0x20) up channel 'Snoop/213023-0000000c' [Aug 18 10:34:31] DEBUG[15020] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[15289] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15289] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[13379] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: pulling 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) [Aug 18 10:34:31] VERBOSE[13379] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 left 'simple_bridge' stasis-bridge <25e1770d-58e8-4da7-94aa-19844c10fa1c> [Aug 18 10:34:31] DEBUG[13379] bridge_channel.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c: 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[13379] bridge_native_rtp.c: Bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[13379] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[13379] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13379] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13379] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[13379] bridge.c: Bridge 25e1770d-58e8-4da7-94aa-19844c10fa1c is already using the new technology. [Aug 18 10:34:31] DEBUG[13379] bridge_channel.c: Bridge is returning 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) to write format slin16 [Aug 18 10:34:31] DEBUG[13379] channel.c: Channel UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0 setting write format path: slin16 -> slin16 [Aug 18 10:34:31] DEBUG[13379] stasis/control.c: robot_212981, 25e1770d-58e8-4da7-94aa-19844c10fa1c: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13379] stasis/app.c: bridge '25e1770d-58e8-4da7-94aa-19844c10fa1c': is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[13379] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) RTP allocated port 11036 [Aug 18 10:34:31] DEBUG[13330] channel.c: Channel 0x7f0c88037560 'Recorder/ARI-0000000a;1' destroying [Aug 18 10:34:31] DEBUG[13330] stasis.c: Destroying topic. name: cache:109/channel:1629282833.91, detail: [Aug 18 10:34:31] DEBUG[13330] stasis.c: Topic 'cache:109/channel:1629282833.91': 0x7f0c880428f0 destroyed [Aug 18 10:34:31] DEBUG[13330] stasis.c: Destroying topic. name: channel:1629282833.91, detail: [Aug 18 10:34:31] DEBUG[13330] stasis.c: Topic 'channel:1629282833.91': 0x7f0c88042730 destroyed [Aug 18 10:34:31] DEBUG[13333] channel.c: Channel 0x7f0cb003b730 'SIP/zvonobot-00000010' destroying [Aug 18 10:34:31] DEBUG[13333] channel.c: Channel 0x7f0c9c0305a0 'Snoop/212981-00000004' destroying [Aug 18 10:34:31] DEBUG[13333] stasis.c: Destroying topic. name: cache:112/channel:1629282833.93, detail: [Aug 18 10:34:31] DEBUG[13333] stasis.c: Topic 'cache:112/channel:1629282833.93': 0x7f0c9c032310 destroyed [Aug 18 10:34:31] DEBUG[13333] stasis.c: Destroying topic. name: channel:1629282833.93, detail: [Aug 18 10:34:31] DEBUG[13333] stasis.c: Topic 'channel:1629282833.93': 0x7f0c9c0321c0 destroyed [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS stop [Aug 18 10:34:31] DEBUG[15003] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:31] DEBUG[15020] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[13341] stasis/control.c: robot_212981: Channel departing bridge [Aug 18 10:34:31] DEBUG[15019] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) ICE creating session 127.0.0.1:11036 (11036) [Aug 18 10:34:31] DEBUG[15003] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15279] channel.c: Channel Recorder/ARI-00000051;2 setting write format path: alaw -> slin [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) ICE create [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) ICE add system candidates [Aug 18 10:34:31] DEBUG[15287] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:31] DEBUG[15287] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) ICE add candidate: 159.65.48.104:11036, 2130706431 [Aug 18 10:34:31] DEBUG[15287] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:31] DEBUG[15287] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:31] DEBUG[15287] res_rtp_asterisk.c: (0x7f0c2001ddd0) ICE add candidate: 10.131.0.10:11036, 2130706431 [Aug 18 10:34:31] DEBUG[15287] rtp_engine.c: RTP instance '0x7f0c2001ddd0' is setup and ready to go [Aug 18 10:34:31] DEBUG[15287] stasis.c: Creating topic. name: channel:robot_213027, detail: [Aug 18 10:34:31] DEBUG[15287] stasis.c: Topic 'channel:robot_213027': 0x7f0c2003ae80 created [Aug 18 10:34:31] DEBUG[15287] stasis.c: Creating topic. name: cache:750/channel:robot_213027, detail: [Aug 18 10:34:31] DEBUG[15287] stasis.c: Topic 'cache:750/channel:robot_213027': 0x7f0c200c2900 created [Aug 18 10:34:31] DEBUG[15290] http.c: match request [ari/channels/212981] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15021] stasis.c: Creating topic. name: channel:1629282871.657, detail: [Aug 18 10:34:31] DEBUG[15021] stasis.c: Topic 'channel:1629282871.657': 0x7f0ca403cf90 created [Aug 18 10:34:31] DEBUG[15021] stasis.c: Creating topic. name: cache:751/channel:1629282871.657, detail: [Aug 18 10:34:31] DEBUG[15021] stasis.c: Topic 'cache:751/channel:1629282871.657': 0x7f0ca402b1e0 created [Aug 18 10:34:31] DEBUG[13341] bridge.c: Waiting for 0x7f0cac04b420(UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0) bridge thread to die. [Aug 18 10:34:31] DEBUG[15291] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15290] http.c: match request [ari/channels/212981] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[15290] http.c: match request [ari/channels/212981] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15289] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c008a30) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9c008a30' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (4) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK437e89d6 Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116864@178.62.121.41", nonce="3f8b6b29", response="43c73b210726c419b446d559637382e6" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266557 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 From: ;tag=as2d7c4d21 To: ;tag=as6fcc16b3 Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" Content-Length: 0 <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7127ccd8;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2d7c4d21 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6fcc16b3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="685bc199" [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 (Checking To) --From tag as2d7c4d21 --To-tag as6fcc16b3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 276fed496e99dd9d468c12053e1e7c08@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #121 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #121)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155085 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890280 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #122 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #122)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116866@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK78caf8e6 Max-Forwards: 70 From: ;tag=as6d2a22e0 To: Contact: Call-ID: 1f0695dc5fa343f27e26b1e410f758d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1712419203 1712419203 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12308 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (1) INVITE - 5 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116836@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 Max-Forwards: 70 From: ;tag=as2d4671b8 To: Contact: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1289209748 1289209748 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3d36e49f4ce645b571e5ba822b9012ed@159.65.48.104:5060' Method: BYE [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS stop [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2400b7d0) ICE RTP transport deallocating [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2400b7d0' [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31f6c45a;received=159.65.48.104 From: ;tag=as4406e1db To: ;tag=as6225b7b2 Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 262 v=0 o=root 51109019 51109019 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 14754 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK31f6c45a;received=159.65.48.104 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4406e1db [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6225b7b2 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 1 [ 45]: o=root 51109019 51109019 IN IP4 178.62.121.41 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 14754 RTP/AVP 0 8 101 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 (Checking To) --From tag as4406e1db --To-tag as6225b7b2 [Aug 18 10:34:31] DEBUG[14038] bridge_channel.c: Setting 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) state from:0 to:1 [Aug 18 10:34:31] DEBUG[14038] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: pulling 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) [Aug 18 10:34:31] VERBOSE[14038] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 left 'simple_bridge' stasis-bridge <051b3352-0990-44a6-b6a2-2bd678146686> [Aug 18 10:34:31] DEBUG[14038] bridge_channel.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686: 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[14038] bridge_native_rtp.c: Bridge '051b3352-0990-44a6-b6a2-2bd678146686' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[14038] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[14038] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14038] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[14038] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[14038] bridge.c: Bridge 051b3352-0990-44a6-b6a2-2bd678146686 is already using the new technology. [Aug 18 10:34:31] DEBUG[15291] http.c: HTTP Request URI is /ari/channels/213009 [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel Recorder/ARI-00000050;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:31] DEBUG[15277] channel.c: Channel Recorder/ARI-00000050;2 setting write format path: alaw -> slin [Aug 18 10:34:31] DEBUG[15290] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Finding handler for channels/212981 [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Finding handler for 212981 [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking channels create: Didn't match 212981 [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15290] res_ari.c: Checking channels externalMedia: Didn't match 212981 [Aug 18 10:34:31] DEBUG[15290] res_ari.c: No explicit handler found for 212981. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15289] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15292] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[13341] stasis/app.c: channel 'robot_212981': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[13341] channel.c: Channel 0x7f0ca40752f0 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:31] DEBUG[15284] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:31] DEBUG[15284] stasis.c: Creating topic. name: channel:1629282871.659, detail: [Aug 18 10:34:31] DEBUG[15284] stasis.c: Topic 'channel:1629282871.659': 0x7f0c24111bc0 created [Aug 18 10:34:31] DEBUG[15284] stasis.c: Creating topic. name: cache:752/channel:1629282871.659, detail: [Aug 18 10:34:31] DEBUG[15284] stasis.c: Topic 'cache:752/channel:1629282871.659': 0x7f0c24051ef0 created [Aug 18 10:34:31] DEBUG[13869] bridge_channel.c: Setting 0x7f0c7c0842d0(Snoop/213023-0000000c) state from:0 to:1 [Aug 18 10:34:31] DEBUG[15292] http.c: HTTP Request URI is /ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Acked pending invite 103 [Aug 18 10:34:31] DEBUG[15294] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15291] http.c: match request [ari/channels/213009] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15291] http.c: match request [ari/channels/213009] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15292] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Stopping retransmission on '2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060' of Request 103: Match Found [Aug 18 10:34:31] DEBUG[15292] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15293] app.c: play_and_record: , /var/spool/asterisk/recording/213025_lNYdPoaZyNgkYDodNHVSYmzebqmBqduB, 'wav' [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP response 200 to standard invite [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Got SDP version 51109019 and unique parts [root 51109019 IN IP4 178.62.121.41] [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing session-level SDP o=root 51109019 51109019 IN IP4 178.62.121.41... OK. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing session-level SDP s=Asterisk PBX 16.18.0... UNSUPPORTED OR FAILED. [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port ''. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing session-level SDP c=IN IP4 178.62.121.41... OK. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found RTP audio format 0 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f0c147e2330 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found RTP audio format 8 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f0c147e2330 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found RTP audio format 101 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401ab20) ICE set role failed; no ice instance [Aug 18 10:34:31] DEBUG[20585] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401ab20) RTCP setting address on RTP instance [Aug 18 10:34:31] VERBOSE[20585] res_rtp_asterisk.c: 0x7f0ca401e710 -- Strict RTP learning after remote address set to: 178.62.121.41:14754 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Peer audio RTP is at port 178.62.121.41:14754 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 0 (0x7f0cb0053f28) from 0x7f0c147e2330 to 0x7f0ca401acf8 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 8 (0x7f0cb0021228) from 0x7f0c147e2330 to 0x7f0ca401acf8 [Aug 18 10:34:31] DEBUG[20585] rtp_engine.c: Copying tx payload mapping 101 (0x7f0cb000c3b8) from 0x7f0c147e2330 to 0x7f0ca401acf8 [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401ab20) RTCP ignoring duplicate property [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: We're settling with these formats: (alaw|ulaw) [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: We have an owner, now see if we need to change this call [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw|ulaw), old nativeformats (alaw) [Aug 18 10:34:31] DEBUG[15291] http.c: match request [ari/channels/213009] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[14038] bridge_channel.c: Bridge is returning 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) to write format slin16 [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:31] DEBUG[13869] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: pulling 0x7f0c7c0842d0(Snoop/213023-0000000c) [Aug 18 10:34:31] VERBOSE[13869] bridge_channel.c: Channel Snoop/213023-0000000c left 'simple_bridge' stasis-bridge <382ca601-8f64-4a7e-bdde-fe8fb07c61bc> [Aug 18 10:34:31] DEBUG[13869] bridge_channel.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: 0x7f0c7c0842d0(Snoop/213023-0000000c) is leaving simple_bridge technology [Aug 18 10:34:31] DEBUG[13869] bridge_native_rtp.c: Bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[13869] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[13869] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13869] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[13869] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[13869] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc is already using the new technology. [Aug 18 10:34:31] DEBUG[13869] bridge_channel.c: Bridge is returning 0x7f0c7c0842d0(Snoop/213023-0000000c) to read format slin [Aug 18 10:34:31] DEBUG[13869] channel.c: Channel Snoop/213023-0000000c setting read format path: slin -> slin [Aug 18 10:34:31] DEBUG[13869] bridge_channel.c: Bridge is returning 0x7f0c7c0842d0(Snoop/213023-0000000c) to write format slin [Aug 18 10:34:31] DEBUG[13869] channel.c: Channel Snoop/213023-0000000c setting write format path: slin -> slin [Aug 18 10:34:31] DEBUG[13869] stasis/control.c: 1629282840.199, 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[13869] stasis/app.c: bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc': is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[13869] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] VERBOSE[15001] res_rtp_asterisk.c: 0x7f0c2c05ce80 -- Strict RTP learning complete - Locking on source address 178.62.121.41:12564 [Aug 18 10:34:31] DEBUG[15291] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15289] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[13785] stasis/control.c: 1629282840.199: Channel departing bridge [Aug 18 10:34:31] DEBUG[13785] bridge.c: Waiting for 0x7f0c7c0842d0(Snoop/213023-0000000c) bridge thread to die. [Aug 18 10:34:31] DEBUG[15294] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:31] DEBUG[15293] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:31] DEBUG[15289] stasis.c: Creating topic. name: bridge:cf85e469-eed6-4be2-b9e9-6396b0eecf75, detail: [Aug 18 10:34:31] DEBUG[15292] http.c: match request [ari/bridges/357a4882-a24d-489f-8ff8-98badd81b2ee] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212981': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[15001] res_rtp_asterisk.c: (0x7f0c2c008d30) RTCP got report of 76 bytes from 178.62.121.41:12565 [Aug 18 10:34:31] DEBUG[14038] channel.c: Channel UnicastRTP/127.0.0.1:50264-0x7f0ca003db80 setting write format path: slin16 -> slin16 [Aug 18 10:34:31] DEBUG[20620] stasis/app.c: channel '212981' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[15289] stasis.c: Topic 'bridge:cf85e469-eed6-4be2-b9e9-6396b0eecf75': 0x7f0c280d94d0 created [Aug 18 10:34:31] DEBUG[15289] stasis.c: Creating topic. name: cache:753/bridge:cf85e469-eed6-4be2-b9e9-6396b0eecf75, detail: [Aug 18 10:34:31] DEBUG[15289] stasis.c: Topic 'cache:753/bridge:cf85e469-eed6-4be2-b9e9-6396b0eecf75': 0x7f0c280a4cc0 created [Aug 18 10:34:31] DEBUG[15289] bridge_native_rtp.c: Bridge 'cf85e469-eed6-4be2-b9e9-6396b0eecf75' can not use native RTP bridge as two channels are required [Aug 18 10:34:31] DEBUG[15289] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:31] DEBUG[15289] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:31] DEBUG[15289] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:31] DEBUG[15289] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:31] DEBUG[15289] bridge.c: Bridge cf85e469-eed6-4be2-b9e9-6396b0eecf75: calling simple_bridge technology constructor [Aug 18 10:34:31] DEBUG[15289] bridge.c: Bridge cf85e469-eed6-4be2-b9e9-6396b0eecf75: calling simple_bridge technology start [Aug 18 10:34:31] DEBUG[15292] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Finding handler for channels/213009 [Aug 18 10:34:31] DEBUG[15295] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Finding handler for bridges/357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:34:31] DEBUG[15295] http.c: HTTP Request URI is /ari/channels/213010/snoop?app=calls_0&spy=in [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[15295] http.c: match request [ari/channels/213010/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15295] http.c: match request [ari/channels/213010/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15289] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:31] DEBUG[15295] http.c: match request [ari/channels/213010/snoop] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15289] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15295] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Finding handler for 357a4882-a24d-489f-8ff8-98badd81b2ee [Aug 18 10:34:31] DEBUG[15292] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15292] res_ari.c: No explicit handler found for 357a4882-a24d-489f-8ff8-98badd81b2ee. Using wildcard bridgeId. [Aug 18 10:34:31] DEBUG[14038] stasis/control.c: robot_213009, 051b3352-0990-44a6-b6a2-2bd678146686: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Finding handler for channels/213010/snoop [Aug 18 10:34:31] DEBUG[15292] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: telling all channels to leave the party [Aug 18 10:34:31] VERBOSE[15293] app.c: x=0, open writing: /var/spool/asterisk/recording/213025_lNYdPoaZyNgkYDodNHVSYmzebqmBqduB format: wav, 0x7f0c38084600 [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Finding handler for channels [Aug 18 10:34:31] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000024 setting read format path: alaw -> alaw [Aug 18 10:34:31] DEBUG[20585] channel.c: Channel SIP/zvonobot-00000024 setting write format path: alaw -> alaw [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401ab20) DTLS - ast_rtp_activate rtp=0x7f0ca401e710 - setup and perform DTLS' [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401e710) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:31] DEBUG[20585] res_rtp_asterisk.c: (0x7f0ca401e710) DTLS perform handshake - ssl = (nil), setup = 0 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:31] VERBOSE[20585] sip/route.c: sip_route_dump: route/path hop: [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Session-Expires: 1800 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Refresher: UAS [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Strict routing enforced for session 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:31] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:31] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821117040@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4341e154 Max-Forwards: 70 From: ;tag=as4406e1db To: ;tag=as6225b7b2 Contact: Call-ID: 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] VERBOSE[13132] dial.c: SIP/zvonobot-00000024 answered [Aug 18 10:34:31] VERBOSE[13132] ari/resource_channels.c: Launching Stasis(calls_0) on SIP/zvonobot-00000024 [Aug 18 10:34:31] DEBUG[15292] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:31] DEBUG[15292] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: queueing action type:13 sub:1001 [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling stasis bridge destructor [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology stop [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 357a4882-a24d-489f-8ff8-98badd81b2ee: calling simple_bridge technology destructor [Aug 18 10:34:31] DEBUG[15292] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:31] DEBUG[15296] http.c: HTTP opening session. Top level [Aug 18 10:34:31] DEBUG[15292] http.c: HTTP closing session. Top level [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Finding handler for 213010 [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channels create: Didn't match 213010 [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channels externalMedia: Didn't match 213010 [Aug 18 10:34:31] DEBUG[15295] res_ari.c: No explicit handler found for 213010. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Finding handler for snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:31] DEBUG[15295] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:31] DEBUG[15296] http.c: HTTP Request URI is /ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: cache:23/channel:212981, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'cache:23/channel:212981': 0x7f0cb00a4450 destroyed [Aug 18 10:34:31] DEBUG[15296] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (3) INVITE - 5 [Aug 18 10:34:31] DEBUG[15294] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:31] DEBUG[15296] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[13785] stasis/app.c: channel '1629282840.199': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[13785] stasis/app.c: channel '1629282840.199' unsubscribed from calls_0 [Aug 18 10:34:31] DEBUG[14038] stasis/app.c: bridge '051b3352-0990-44a6-b6a2-2bd678146686': is 3 interested in calls_0 [Aug 18 10:34:31] DEBUG[13132] stasis/app.c: Channel '213000' is 2 interested in calls_0 [Aug 18 10:34:31] DEBUG[14038] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[13993] stasis/control.c: robot_213009: Channel departing bridge [Aug 18 10:34:31] DEBUG[15296] http.c: match request [ari/bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:31] DEBUG[20620] stasis.c: Destroying topic. name: channel:212981, detail: [Aug 18 10:34:31] DEBUG[20620] stasis.c: Topic 'channel:212981': 0x7f0cb00299b0 destroyed [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:31] DEBUG[13785] channel.c: Channel 0x7f0c180b81a0 'Snoop/213023-0000000c' hanging up. Refs: 3 [Aug 18 10:34:31] VERBOSE[13132] res_rtp_asterisk.c: 0x7f0ca401e710 -- Strict RTP switching to RTP target address 178.62.121.41:14754 as source [Aug 18 10:34:31] DEBUG[13132] chan_sip.c: Oooh, format changed to ulaw [Aug 18 10:34:31] DEBUG[13132] channel.c: Channel SIP/zvonobot-00000024 setting read format path: ulaw -> alaw [Aug 18 10:34:31] DEBUG[13993] bridge.c: Waiting for 0x7f0c18085a90(UnicastRTP/127.0.0.1:50264-0x7f0ca003db80) bridge thread to die. [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:31] DEBUG[15294] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:31] DEBUG[13993] stasis/app.c: channel 'robot_213009': is 1 interested in calls_0 [Aug 18 10:34:31] DEBUG[13993] channel.c: Channel 0x7f0ca00dd400 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80' hanging up. Refs: 2 [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:31] DEBUG[15294] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:31] DEBUG[13132] channel.c: Channel SIP/zvonobot-00000024 setting write format path: alaw -> ulaw [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:31] DEBUG[15296] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Finding handler for 213009 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #85 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[15294] http.c: Match made with [ari] [Aug 18 10:34:31] DEBUG[15175] channel.c: Channel 0x7f0cb015d770 'SIP/zvonobot-000000f1' allocated [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[15175] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[14843] chan_sip.c: Hangup call SIP/zvonobot-000000c1, SIP callid 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 [Aug 18 10:34:31] DEBUG[15198] channel.c: Channel 0x7f0c2c0edb40 'Recorder/ARI-00000058;1' allocated [Aug 18 10:34:31] DEBUG[15198] stasis.c: Creating topic. name: channel:1629282871.660, detail: [Aug 18 10:34:31] DEBUG[15198] stasis.c: Topic 'channel:1629282871.660': 0x7f0c2c0a9170 created [Aug 18 10:34:31] DEBUG[15198] stasis.c: Creating topic. name: cache:754/channel:1629282871.660, detail: [Aug 18 10:34:31] DEBUG[15198] stasis.c: Topic 'cache:754/channel:1629282871.660': 0x7f0c2c015950 created [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #85)) [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731456 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking channels create: Didn't match 213009 [Aug 18 10:34:31] DEBUG[15208] channel.c: Channel 0x7f0c7806bc10 'Recorder/ARI-00000057;1' allocated [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15291] res_ari.c: Checking channels externalMedia: Didn't match 213009 [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[14843] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:31] DEBUG[14843] res_rtp_asterisk.c: (0x7f0c400bd8b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14843] res_rtp_asterisk.c: (0x7f0c400bd8b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:31] DEBUG[14843] channel.c: Channel 0x7f0c400c7ad0 'SIP/zvonobot-000000c1' destroying [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Finding handler for bridges/382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15291] res_ari.c: No explicit handler found for 213009. Using wildcard channelId. [Aug 18 10:34:31] DEBUG[15208] stasis.c: Creating topic. name: channel:1629282871.661, detail: [Aug 18 10:34:31] DEBUG[15208] stasis.c: Topic 'channel:1629282871.661': 0x7f0c78052f90 created [Aug 18 10:34:31] DEBUG[15208] stasis.c: Creating topic. name: cache:755/channel:1629282871.661, detail: [Aug 18 10:34:31] DEBUG[15208] stasis.c: Topic 'cache:755/channel:1629282871.661': 0x7f0c78016b50 created [Aug 18 10:34:31] DEBUG[15294] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15294] res_ari.c: Finding handler for bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #80 (5) INVITE - 5 [Aug 18 10:34:31] DEBUG[15294] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 538 [Aug 18 10:34:31] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 538 [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #80)) [Aug 18 10:34:31] DEBUG[15294] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Finding handler for 382ca601-8f64-4a7e-bdde-fe8fb07c61bc [Aug 18 10:34:31] DEBUG[15296] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:31] DEBUG[15296] res_ari.c: No explicit handler found for 382ca601-8f64-4a7e-bdde-fe8fb07c61bc. Using wildcard bridgeId. [Aug 18 10:34:31] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116858@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3839516f Max-Forwards: 70 From: ;tag=as2e21b584 To: Contact: Call-ID: 21c4169c011e5d806119facd5d9285fb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 984372102 984372102 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17486 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:31] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:31] DEBUG[15179] channel.c: Channel 0x7f0cac0185f0 'SIP/zvonobot-000000f2' allocated [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:31] DEBUG[15179] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:31] DEBUG[15296] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: telling all channels to leave the party [Aug 18 10:34:31] DEBUG[15296] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:31] DEBUG[15296] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: queueing action type:13 sub:1001 [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.662, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.662': 0x7f0c300413e0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Creating topic. name: cache:756/channel:1629282871.662, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:756/channel:1629282871.662': 0x7f0c300926b0 created [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: cache:756/channel:1629282871.662, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'cache:756/channel:1629282871.662': 0x7f0c300926b0 destroyed [Aug 18 10:34:31] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.662, detail: [Aug 18 10:34:31] DEBUG[20545] stasis.c: Topic 'channel:1629282871.662': 0x7f0c300413e0 destroyed [Aug 18 10:34:31] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000010', '', 'Stasis', 'calls_0', 43, 34, 'ANSWERED', 3, '', '212981', '')] [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: calling stasis bridge destructor [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: calling simple_bridge technology stop [Aug 18 10:34:31] DEBUG[20534] bridge.c: Bridge 382ca601-8f64-4a7e-bdde-fe8fb07c61bc: calling simple_bridge technology destructor [Aug 18 10:34:31] DEBUG[15296] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282871.663, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282871.663': 0x7f0c300413e0 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: cache:757/channel:1629282871.663, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:757/channel:1629282871.663': 0x7f0c300926b0 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: cache:757/channel:1629282871.663, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:757/channel:1629282871.663': 0x7f0c300926b0 destroyed [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282871.663, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282871.663': 0x7f0c300413e0 destroyed [Aug 18 10:34:32] DEBUG[15296] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: bridge '357a4882-a24d-489f-8ff8-98badd81b2ee': is 0 interested in calls_0 [Aug 18 10:34:31] DEBUG[15294] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: bridge '357a4882-a24d-489f-8ff8-98badd81b2ee' unsubscribed from calls_0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:32] DEBUG[15294] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15294] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: cache:191/bridge:357a4882-a24d-489f-8ff8-98badd81b2ee, detail: [Aug 18 10:34:32] DEBUG[15207] channel.c: Channel 0x7f0c7c080580 'SIP/zvonobot-000000f3' allocated [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15207] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'cache:191/bridge:357a4882-a24d-489f-8ff8-98badd81b2ee': 0x7f0c3c043610 destroyed [Aug 18 10:34:32] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'Snoop/212981-00000004', 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0', 'Stasis', 'calls_0', 36, 36, 'ANSWERED', 3, '', '1629282833.93', '')] [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: bridge:357a4882-a24d-489f-8ff8-98badd81b2ee, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'bridge:357a4882-a24d-489f-8ff8-98badd81b2ee': 0x7f0c3c060df0 destroyed [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:32] DEBUG[15294] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15294] stasis.c: Creating topic. name: bridge:a83ab563-1924-459b-b74d-9b0306807c5a, detail: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Session timer started: 57 - 2224131f10bb1004792fcd5d3ee3bb0a@159.65.48.104:5060 1768000ms [Aug 18 10:34:32] DEBUG[15175] res_stasis.c: calls_0: Subscribing to 213206 [Aug 18 10:34:32] DEBUG[15175] stasis/app.c: Channel '213206' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15175] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 477 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 477 [Aug 18 10:34:32] DEBUG[15175] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15179] res_stasis.c: calls_0: Subscribing to 213205 [Aug 18 10:34:32] DEBUG[15179] stasis/app.c: Channel '213205' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15298] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Outgoing Call for 79821116834 [Aug 18 10:34:32] DEBUG[15207] res_stasis.c: calls_0: Subscribing to 213211 [Aug 18 10:34:32] DEBUG[15207] stasis/app.c: Channel '213211' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Outgoing Call for 79821116835 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15299] chan_sip.c: Audio is at 10426 [Aug 18 10:34:32] VERBOSE[15299] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15299] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[15299] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15298] http.c: HTTP Request URI is /ari/channels/213215?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116825&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] DEBUG[15179] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] VERBOSE[15300] chan_sip.c: Audio is at 17162 [Aug 18 10:34:32] VERBOSE[15300] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] DEBUG[15303] http.c: HTTP opening session. Top level [Aug 18 10:34:32] VERBOSE[15300] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Initializing initreq for method INVITE - callid 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116834@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 3 [ 52]: From: ;tag=as2f225ac7 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 6 [ 60]: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15299] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116834@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc Max-Forwards: 70 From: ;tag=as2f225ac7 To: Contact: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637714792 637714792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10426 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #119 [Aug 18 10:34:32] DEBUG[15299] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:32] VERBOSE[15299] dial.c: Called zvonobot/79821116834 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Outgoing Call for 79821116829 [Aug 18 10:34:32] DEBUG[15207] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[15207] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15303] http.c: HTTP Request URI is /ari/channels/213214?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116826&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15179] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15294] stasis.c: Topic 'bridge:a83ab563-1924-459b-b74d-9b0306807c5a': 0x7f0c40007560 created [Aug 18 10:34:32] DEBUG[15294] stasis.c: Creating topic. name: cache:758/bridge:a83ab563-1924-459b-b74d-9b0306807c5a, detail: [Aug 18 10:34:32] DEBUG[15294] stasis.c: Topic 'cache:758/bridge:a83ab563-1924-459b-b74d-9b0306807c5a': 0x7f0c4005a1f0 created [Aug 18 10:34:32] DEBUG[15294] bridge_native_rtp.c: Bridge 'a83ab563-1924-459b-b74d-9b0306807c5a' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] DEBUG[15294] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15294] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15294] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:32] DEBUG[15294] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15294] bridge.c: Bridge a83ab563-1924-459b-b74d-9b0306807c5a: calling simple_bridge technology constructor [Aug 18 10:34:32] DEBUG[15294] bridge.c: Bridge a83ab563-1924-459b-b74d-9b0306807c5a: calling simple_bridge technology start [Aug 18 10:34:32] DEBUG[15294] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] VERBOSE[15300] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15294] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15298] http.c: match request [ari/channels/213215] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Initializing initreq for method INVITE - callid 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116835@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 3 [ 52]: From: ;tag=as2e8baab5 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 6 [ 60]: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15300] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116835@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 Max-Forwards: 70 From: ;tag=as2e8baab5 To: Contact: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1705768465 1705768465 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17162 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #135 [Aug 18 10:34:32] DEBUG[15300] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986;received=159.65.48.104 From: ;tag=as0bc44772 To: ;tag=as1488a4e7 Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b319825" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[15298] http.c: match request [ari/channels/213215] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15305] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15307] http.c: HTTP opening session. Top level [Aug 18 10:34:32] VERBOSE[15300] dial.c: Called zvonobot/79821116835 [Aug 18 10:34:32] DEBUG[15298] http.c: match request [ari/channels/213215] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[15305] http.c: HTTP Request URI is /ari/channels/213025/snoop?app=calls_0&spy=in [Aug 18 10:34:32] DEBUG[15015] res_rtp_asterisk.c: (0x7f0c8c020490) RTCP got report of 76 bytes from 178.62.121.41:17185 [Aug 18 10:34:32] DEBUG[15298] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0bc44772 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as1488a4e7 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b319825" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 (Checking To) --From tag as0bc44772 --To-tag as1488a4e7 [Aug 18 10:34:32] DEBUG[15305] http.c: match request [ari/channels/213025/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15305] http.c: match request [ari/channels/213025/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15307] http.c: HTTP Request URI is /ari/channels/213216?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116824&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15298] http.c: HTTP consuming request body [Aug 18 10:34:32] VERBOSE[15015] res_rtp_asterisk.c: 0x7f0c8c0246a0 -- Strict RTP learning complete - Locking on source address 178.62.121.41:17184 [Aug 18 10:34:32] DEBUG[15305] http.c: match request [ari/channels/213025/snoop] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15305] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Finding handler for channels/213215 [Aug 18 10:34:32] DEBUG[13292] chan_sip.c: Hangup call SIP/zvonobot-0000003b, SIP callid 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[13292] res_rtp_asterisk.c: (0x7f0c7c01e650) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[13292] res_rtp_asterisk.c: (0x7f0c7c01e650) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] VERBOSE[13292] chan_sip.c: Scheduling destruction of SIP dialog '3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060' in 6400 ms (Method: INVITE) [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15307] http.c: match request [ari/channels/213216] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15307] http.c: match request [ari/channels/213216] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15307] http.c: match request [ari/channels/213216] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15307] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15307] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Finding handler for channels/213216 [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Finding handler for 213216 [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking channels create: Didn't match 213216 [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Finding handler for channels/213025/snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15307] res_ari.c: Checking channels externalMedia: Didn't match 213216 [Aug 18 10:34:32] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 19 instead [Aug 18 10:34:32] DEBUG[15303] http.c: match request [ari/channels/213214] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Finding handler for 213025 [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channels create: Didn't match 213025 [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channels externalMedia: Didn't match 213025 [Aug 18 10:34:32] DEBUG[15305] res_ari.c: No explicit handler found for 213025. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Finding handler for snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:32] DEBUG[15305] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213159': is 0 interested in calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213159' unsubscribed from calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: cache:523/channel:213159, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'cache:523/channel:213159': 0x7f0c400ca430 destroyed [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: channel:213159, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'channel:213159': 0x7f0c400c6fe0 destroyed [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc': is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: bridge '382ca601-8f64-4a7e-bdde-fe8fb07c61bc' unsubscribed from calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: cache:224/bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc, detail: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #85 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Stopping retransmission on '0b26716f0c119a15755932c124bb341d@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4228b986 Max-Forwards: 70 From: ;tag=as0bc44772 To: ;tag=as1488a4e7 Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[15303] http.c: match request [ari/channels/213214] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[13292] chan_sip.c: Strict routing enforced for session 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[13292] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 18 10:34:32] DEBUG[13292] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:32] DEBUG[13292] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:32] VERBOSE[13292] chan_sip.c: set_destination: set destination to 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[13292] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: BYE sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK204acf50 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41:5060", nonce="0de4ceb7", response="5f946298dc53c0cdee5695c18ecb459f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15301] chan_sip.c: Audio is at 13902 [Aug 18 10:34:32] VERBOSE[15301] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15301] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[15301] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[13292] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22 [Aug 18 10:34:32] DEBUG[13292] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'cache:224/bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc': 0x7f0ca402b040 destroyed [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Finding handler for 213215 [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking channels create: Didn't match 213215 [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15298] res_ari.c: Checking channels externalMedia: Didn't match 213215 [Aug 18 10:34:32] DEBUG[15298] res_ari.c: No explicit handler found for 213215. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15307] res_ari.c: No explicit handler found for 213216. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15310] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'bridge:382ca601-8f64-4a7e-bdde-fe8fb07c61bc': 0x7f0ca40044e0 destroyed [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[15303] http.c: match request [ari/channels/213214] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15310] http.c: HTTP Request URI is /ari/channels/213217?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116823&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15315] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Initializing initreq for method INVITE - callid 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116829@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 3 [ 52]: From: ;tag=as5c283596 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 6 [ 60]: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15301] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116829@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 Max-Forwards: 70 From: ;tag=as5c283596 To: Contact: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2033823795 2033823795 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #117 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Audio is at 18924 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2baf6b12 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116862@178.62.121.41", nonce="0b319825", response="6a20eac9692b8e193e543b5bdc77a717" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK437e89d6 Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116864@178.62.121.41", nonce="3f8b6b29", response="43c73b210726c419b446d559637382e6" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266557 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230767 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116836@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 Max-Forwards: 70 From: ;tag=as2d4671b8 To: Contact: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1289209748 1289209748 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2e8d66bf05763a492c28a0cb7f31a7f8@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c400bd8b0) DTLS stop [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c400bd8b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c400bd8b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c400bd8b0) ICE RTP transport deallocating [Aug 18 10:34:32] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c400bd8b0' [Aug 18 10:34:32] DEBUG[20545] cdr.c: Finalized CDR for Snoop/213009-0000000e - start 1629282843.045775 answer 1629282843.045775 end 1629282871.854191 dur 28.808 bill 28.808 dispo ANSWERED [Aug 18 10:34:32] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:32] DEBUG[15303] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15303] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Finding handler for channels/213214 [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Finding handler for 213214 [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking channels create: Didn't match 213214 [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15303] res_ari.c: Checking channels externalMedia: Didn't match 213214 [Aug 18 10:34:32] DEBUG[15303] res_ari.c: No explicit handler found for 213214. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15310] http.c: match request [ari/channels/213217] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15310] http.c: match request [ari/channels/213217] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15310] http.c: match request [ari/channels/213217] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15310] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15199] channel.c: Channel 0x7f0c280d41a0 'SIP/zvonobot-000000f4' allocated [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15199] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[15301] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15310] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15315] http.c: HTTP Request URI is /ari/channels/213218?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116822&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15316] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Session timer stopped: 39 - 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #134 (6) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 6400 ms (t1 100 ms (Retrans id #134)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #6 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 688963152 688963152 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17132 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #126 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #126)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116857@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1ccd9b24 Max-Forwards: 70 From: ;tag=as329379c0 To: Contact: Call-ID: 761f2e08240929491af55d7f51e15ac6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1485885243 1485885243 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10852 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15316] http.c: HTTP Request URI is /ari/channels/213219?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116821&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15199] res_stasis.c: calls_0: Subscribing to 213207 [Aug 18 10:34:32] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTP audio difference is 896, ms is 132 [Aug 18 10:34:32] DEBUG[15316] http.c: match request [ari/channels/213219] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15317] http.c: HTTP opening session. Top level [Aug 18 10:34:32] VERBOSE[15301] dial.c: Called zvonobot/79821116829 [Aug 18 10:34:32] DEBUG[15317] http.c: HTTP Request URI is /ari/channels/213221?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116819&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15042] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:32] DEBUG[15199] stasis/app.c: Channel '213207' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Finding handler for channels/213217 [Aug 18 10:34:32] DEBUG[15213] channel.c: Channel 0x7f0c8806f020 'Recorder/ARI-00000059;1' allocated [Aug 18 10:34:32] DEBUG[15213] stasis.c: Creating topic. name: channel:1629282872.664, detail: [Aug 18 10:34:32] DEBUG[15213] stasis.c: Topic 'channel:1629282872.664': 0x7f0c8809e130 created [Aug 18 10:34:32] DEBUG[15213] stasis.c: Creating topic. name: cache:759/channel:1629282872.664, detail: [Aug 18 10:34:32] DEBUG[15213] stasis.c: Topic 'cache:759/channel:1629282872.664': 0x7f0c880340d0 created [Aug 18 10:34:32] DEBUG[15042] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[13698] channel.c: Channel 0x7f0c94066640 'Recorder/ARI-0000001e;1' destroying [Aug 18 10:34:32] DEBUG[13698] stasis.c: Destroying topic. name: cache:207/channel:1629282838.174, detail: [Aug 18 10:34:32] DEBUG[13698] stasis.c: Topic 'cache:207/channel:1629282838.174': 0x7f0c940682b0 destroyed [Aug 18 10:34:32] DEBUG[13698] stasis.c: Destroying topic. name: channel:1629282838.174, detail: [Aug 18 10:34:32] DEBUG[13698] stasis.c: Topic 'channel:1629282838.174': 0x7f0c94064c70 destroyed [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Finding handler for channels [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 From: ;tag=as76ba9cfd To: ;tag=as041dcfd6 Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK182a1bd0;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as76ba9cfd [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as041dcfd6 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2621991b" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 (Checking To) --From tag as76ba9cfd --To-tag as041dcfd6 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 18b439a22ac69e570c5898a94043ad6c@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #119 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #119)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116834@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc Max-Forwards: 70 From: ;tag=as2f225ac7 To: Contact: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637714792 637714792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10426 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6380ms with no response [Aug 18 10:34:32] WARNING[20585] chan_sip.c: Hanging up call 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[14912] channel.c: Channel 0x7f0c100480d0 'SIP/zvonobot-000000c8' hanging up. Refs: 2 [Aug 18 10:34:32] DEBUG[15199] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[15316] http.c: match request [ari/channels/213219] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15199] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15316] http.c: match request [ari/channels/213219] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15317] http.c: match request [ari/channels/213221] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282872.665, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.665': 0x7f0c300413e0 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: cache:760/channel:1629282872.665, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:760/channel:1629282872.665': 0x7f0c300926b0 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: cache:760/channel:1629282872.665, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:760/channel:1629282872.665': 0x7f0c300926b0 destroyed [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282872.665, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.665': 0x7f0c300413e0 destroyed [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Outgoing Call for 79821116833 [Aug 18 10:34:32] DEBUG[15319] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15316] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116835@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 Max-Forwards: 70 From: ;tag=as2e8baab5 To: Contact: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1705768465 1705768465 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17162 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Aug 18 10:34:32] DEBUG[15316] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15317] http.c: match request [ari/channels/213221] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Finding handler for 213217 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15315] http.c: match request [ari/channels/213218] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15317] http.c: match request [ari/channels/213221] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking channels create: Didn't match 213217 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:32] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:23', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000c1', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213159', '')] [Aug 18 10:34:32] DEBUG[15319] http.c: HTTP Request URI is /ari/channels/213222?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116818&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: Checking channels externalMedia: Didn't match 213217 [Aug 18 10:34:32] DEBUG[15310] res_ari.c: No explicit handler found for 213217. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15320] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Finding handler for channels/213219 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 From: ;tag=as2ff9bb68 To: ;tag=as6dc6d37f Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK779b2776;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2ff9bb68 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6dc6d37f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="697e6bd0" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[15315] http.c: match request [ari/channels/213218] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15317] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15059] channel.c: Channel 0x7f0c74091b80 'Recorder/ARI-00000053;1' allocated [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15059] stasis.c: Creating topic. name: channel:1629282872.666, detail: [Aug 18 10:34:32] DEBUG[15201] channel.c: Channel 0x7f0c300b8b50 'SIP/zvonobot-000000f5' allocated [Aug 18 10:34:32] DEBUG[15059] stasis.c: Topic 'channel:1629282872.666': 0x7f0c74032b40 created [Aug 18 10:34:32] DEBUG[15059] stasis.c: Creating topic. name: cache:761/channel:1629282872.666, detail: [Aug 18 10:34:32] DEBUG[15059] stasis.c: Topic 'cache:761/channel:1629282872.666': 0x7f0c74044e20 created [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 (Checking To) --From tag as2ff9bb68 --To-tag as6dc6d37f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 11dc081f3c39bcc773246e4d11d4ee31@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15319] http.c: match request [ari/channels/213222] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[15315] http.c: match request [ari/channels/213218] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15321] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #160 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] http.c: HTTP Request URI is /ari/channels/213223?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116817&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #160)) [Aug 18 10:34:32] DEBUG[15320] http.c: HTTP Request URI is /ari/channels/213220?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116820&callerId=74950493843 [Aug 18 10:34:32] DEBUG[15319] http.c: match request [ari/channels/213222] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15315] http.c: Match made with [ari] [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15319] http.c: match request [ari/channels/213222] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: Allocating new SIP dialog for 1682582a6011ced77abab78973c2084e@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15298] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c7c0bb910' [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) RTP allocated port 17982 [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE creating session 0.0.0.0:17982 (17982) [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE create [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE add system candidates [Aug 18 10:34:32] DEBUG[15298] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15298] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE add candidate: 159.65.48.104:17982, 2130706431 [Aug 18 10:34:32] DEBUG[15298] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15298] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE add candidate: 10.131.0.10:17982, 2130706431 [Aug 18 10:34:32] DEBUG[15298] rtp_engine.c: RTP instance '0x7f0c7c0bb910' is setup and ready to go [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) ICE stopped [Aug 18 10:34:32] DEBUG[15298] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15298] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15298] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15298] res_rtp_asterisk.c: (0x7f0c7c0bb910) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15298] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15298] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] DEBUG[15321] http.c: match request [ari/channels/213223] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15201] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[15320] http.c: match request [ari/channels/213220] with handler [httpstatus] len 10 [Aug 18 10:34:32] VERBOSE[15318] chan_sip.c: Audio is at 14362 [Aug 18 10:34:32] DEBUG[15320] http.c: match request [ari/channels/213220] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15315] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15298] chan_sip.c: SIP call-id changed from '1682582a6011ced77abab78973c2084e@127.0.1.1:5060' to '4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15317] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15320] http.c: match request [ari/channels/213220] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15319] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:32] DEBUG[15321] http.c: match request [ari/channels/213223] with handler [phoneprov] len 9 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #22 (1) BYE - 8 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #22)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: BYE sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK204acf50 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41:5060", nonce="0de4ceb7", response="5f946298dc53c0cdee5695c18ecb459f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #117 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #117)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116829@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 Max-Forwards: 70 From: ;tag=as5c283596 To: Contact: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2033823795 2033823795 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:32] DEBUG[15320] http.c: Match made with [ari] [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2baf6b12 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116862@178.62.121.41", nonce="0b319825", response="6a20eac9692b8e193e543b5bdc77a717" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Finding handler for channels/213221 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Finding handler for 213219 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking channels create: Didn't match 213219 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15321] http.c: match request [ari/channels/213223] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15320] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[15318] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15316] res_ari.c: Checking channels externalMedia: Didn't match 213219 [Aug 18 10:34:32] DEBUG[15316] res_ari.c: No explicit handler found for 213219. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: Allocating new SIP dialog for 0559912372a524e21b48ac7d688fbac9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15303] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c06e200' [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) RTP allocated port 15260 [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE creating session 0.0.0.0:15260 (15260) [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE create [Aug 18 10:34:32] DEBUG[15298] stasis.c: Creating topic. name: channel:213215, detail: [Aug 18 10:34:32] DEBUG[15298] stasis.c: Topic 'channel:213215': 0x7f0c7c0bd9f0 created [Aug 18 10:34:32] DEBUG[15298] stasis.c: Creating topic. name: cache:762/channel:213215, detail: [Aug 18 10:34:32] DEBUG[15298] stasis.c: Topic 'cache:762/channel:213215': 0x7f0c7c0099f0 created [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Finding handler for channels/213218 [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Finding handler for 213218 [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking channels create: Didn't match 213218 [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15315] res_ari.c: Checking channels externalMedia: Didn't match 213218 [Aug 18 10:34:32] DEBUG[15315] res_ari.c: No explicit handler found for 213218. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:32] DEBUG[15319] http.c: HTTP consuming request body [Aug 18 10:34:32] VERBOSE[15318] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Finding handler for channels/213220 [Aug 18 10:34:32] DEBUG[15321] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE add system candidates [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Finding handler for channels/213222 [Aug 18 10:34:32] DEBUG[15201] res_stasis.c: calls_0: Subscribing to 213209 [Aug 18 10:34:32] DEBUG[15201] stasis/app.c: Channel '213209' is 1 interested in calls_0 [Aug 18 10:34:32] VERBOSE[15318] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Outgoing Call for 79821116831 [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15201] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15303] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15201] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] http.c: HTTP consuming request body [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15303] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE add candidate: 159.65.48.104:15260, 2130706431 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[14799] res_rtp_asterisk.c: (0x7f0ca000a6f0) RTCP got report of 76 bytes from 178.62.121.41:15547 [Aug 18 10:34:32] DEBUG[15303] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15303] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Finding handler for channels/213223 [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Finding handler for 213223 [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking channels create: Didn't match 213223 [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15321] res_ari.c: Checking channels externalMedia: Didn't match 213223 [Aug 18 10:34:32] DEBUG[15321] res_ari.c: No explicit handler found for 213223. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Finding handler for 213222 [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE add candidate: 10.131.0.10:15260, 2130706431 [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking channels create: Didn't match 213222 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 From: ;tag=as2eb39fa6 To: ;tag=as7b46504f Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Initializing initreq for method INVITE - callid 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116833@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 3 [ 52]: From: ;tag=as235fcd7f [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 6 [ 60]: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15318] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116833@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d Max-Forwards: 70 From: ;tag=as235fcd7f To: Contact: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1532262111 1532262111 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14362 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25 [Aug 18 10:34:32] DEBUG[15318] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4247e838;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[15303] rtp_engine.c: RTP instance '0x7f0c8c06e200' is setup and ready to go [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Finding handler for 213220 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as2eb39fa6 [Aug 18 10:34:32] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000c8 - start 1629282865.667651 answer 0.000000 end 1629282872.205316 dur 6.537 bill 1629282872.205 dispo NO ANSWER [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7b46504f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15319] res_ari.c: Checking channels externalMedia: Didn't match 213222 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] VERBOSE[15318] dial.c: Called zvonobot/79821116833 [Aug 18 10:34:32] DEBUG[15319] res_ari.c: No explicit handler found for 213222. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e3ccab" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 (Checking To) --From tag as2eb39fa6 --To-tag as7b46504f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 31084e6149d402b41e86a7dd14209045@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #23 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #23)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #115 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #115)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116870@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK48570e59 Max-Forwards: 70 From: ;tag=as36bdbaac To: Contact: Call-ID: 5a82d08913ae08152dcd64d1249c7859@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 136960714 136960714 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14652 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #125 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #125)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #119 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #119)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116834@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc Max-Forwards: 70 From: ;tag=as2f225ac7 To: Contact: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637714792 637714792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10426 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116835@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 Max-Forwards: 70 From: ;tag=as2e8baab5 To: Contact: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1705768465 1705768465 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17162 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) ICE stopped [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking channels create: Didn't match 213220 [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: Allocating new SIP dialog for 6419d3802a9e0d3b421c8618499d3098@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15307] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9405fa90' [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) RTP allocated port 14390 [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE creating session 0.0.0.0:14390 (14390) [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE create [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE add system candidates [Aug 18 10:34:32] DEBUG[15307] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15307] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE add candidate: 159.65.48.104:14390, 2130706431 [Aug 18 10:34:32] DEBUG[15307] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15307] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Finding handler for 213221 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking channels create: Didn't match 213221 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15317] res_ari.c: Checking channels externalMedia: Didn't match 213221 [Aug 18 10:34:32] DEBUG[15317] res_ari.c: No explicit handler found for 213221. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029194 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK437e89d6 Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116864@178.62.121.41", nonce="3f8b6b29", response="43c73b210726c419b446d559637382e6" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266557 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE add candidate: 10.131.0.10:14390, 2130706431 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK73a7e9a8 Max-Forwards: 70 From: ;tag=as67bc1c22 To: ;tag=as22df0306 Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="35308dad", response="75074e8706ca6826e8bd1740d88e8740" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK73a7e9a8 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as67bc1c22 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as22df0306 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15303] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] DEBUG[15320] res_ari.c: Checking channels externalMedia: Didn't match 213220 [Aug 18 10:34:32] DEBUG[15320] res_ari.c: No explicit handler found for 213220. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: Allocating new SIP dialog for 5b17278357c8e12d0780b19c763a92f8@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15307] rtp_engine.c: RTP instance '0x7f0c9405fa90' is setup and ready to go [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) ICE stopped [Aug 18 10:34:32] DEBUG[15307] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15307] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15307] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="35308dad", response="75074e8706ca6826e8bd1740d88e8740" [Aug 18 10:34:32] DEBUG[15303] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15316] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c0e5ff0' [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) RTP allocated port 15514 [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE creating session 0.0.0.0:15514 (15514) [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE create [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE add system candidates [Aug 18 10:34:32] DEBUG[15316] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15316] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE add candidate: 159.65.48.104:15514, 2130706431 [Aug 18 10:34:32] DEBUG[15316] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15316] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE add candidate: 10.131.0.10:15514, 2130706431 [Aug 18 10:34:32] DEBUG[15316] rtp_engine.c: RTP instance '0x7f0c9c0e5ff0' is setup and ready to go [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) ICE stopped [Aug 18 10:34:32] DEBUG[15316] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15316] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15316] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15316] res_rtp_asterisk.c: (0x7f0c9c0e5ff0) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15316] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15316] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15316] chan_sip.c: SIP call-id changed from '5b17278357c8e12d0780b19c763a92f8@127.0.1.1:5060' to '315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15316] stasis.c: Creating topic. name: channel:213219, detail: [Aug 18 10:34:32] DEBUG[15316] stasis.c: Topic 'channel:213219': 0x7f0c9c009560 created [Aug 18 10:34:32] DEBUG[15316] stasis.c: Creating topic. name: cache:763/channel:213219, detail: [Aug 18 10:34:32] DEBUG[15316] stasis.c: Topic 'cache:763/channel:213219': 0x7f0c9c00b4a0 created [Aug 18 10:34:32] DEBUG[15303] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 (Checking From) --From tag as67bc1c22 --To-tag as22df0306 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:32] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15307] res_rtp_asterisk.c: (0x7f0c9405fa90) RTCP setup on RTP instance [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15303] res_rtp_asterisk.c: (0x7f0c8c06e200) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15322] chan_sip.c: Audio is at 19488 [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: Allocating new SIP dialog for 2531f6200f4aca5350525ec934573a9c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] VERBOSE[15303] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15303] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[15307] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15307] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: Allocating new SIP dialog for 7acb35612eb2724741b79a8a4c620ad7@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15315] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca806d6c0' [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) RTP allocated port 14728 [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE creating session 0.0.0.0:14728 (14728) [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE create [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE add system candidates [Aug 18 10:34:32] DEBUG[15315] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15315] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE add candidate: 159.65.48.104:14728, 2130706431 [Aug 18 10:34:32] DEBUG[15315] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15315] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE add candidate: 10.131.0.10:14728, 2130706431 [Aug 18 10:34:32] DEBUG[15315] rtp_engine.c: RTP instance '0x7f0ca806d6c0' is setup and ready to go [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) ICE stopped [Aug 18 10:34:32] DEBUG[15315] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15315] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15315] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15303] chan_sip.c: SIP call-id changed from '0559912372a524e21b48ac7d688fbac9@127.0.1.1:5060' to '01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15303] stasis.c: Creating topic. name: channel:213214, detail: [Aug 18 10:34:32] DEBUG[15303] stasis.c: Topic 'channel:213214': 0x7f0c8c03ced0 created [Aug 18 10:34:32] DEBUG[15303] stasis.c: Creating topic. name: cache:764/channel:213214, detail: [Aug 18 10:34:32] DEBUG[15303] stasis.c: Topic 'cache:764/channel:213214': 0x7f0c8c10b0b0 created [Aug 18 10:34:32] DEBUG[15315] res_rtp_asterisk.c: (0x7f0ca806d6c0) RTCP setup on RTP instance [Aug 18 10:34:32] DEBUG[15310] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c900ac120' [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) RTP allocated port 11192 [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE creating session 0.0.0.0:11192 (11192) [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE create [Aug 18 10:34:32] DEBUG[15307] chan_sip.c: SIP call-id changed from '6419d3802a9e0d3b421c8618499d3098@127.0.1.1:5060' to '2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15307] stasis.c: Creating topic. name: channel:213216, detail: [Aug 18 10:34:32] DEBUG[15307] stasis.c: Topic 'channel:213216': 0x7f0c94067a30 created [Aug 18 10:34:32] DEBUG[15307] stasis.c: Creating topic. name: cache:765/channel:213216, detail: [Aug 18 10:34:32] DEBUG[15307] stasis.c: Topic 'cache:765/channel:213216': 0x7f0c940ccb60 created [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE add system candidates [Aug 18 10:34:32] VERBOSE[15322] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15315] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15310] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3c031e30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: Allocating new SIP dialog for 73829489627e34e909638d6027fbd19c@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15321] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac01e130' [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) RTP allocated port 13638 [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE creating session 0.0.0.0:13638 (13638) [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE create [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add system candidates [Aug 18 10:34:32] DEBUG[15321] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15321] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 159.65.48.104:13638, 2130706431 [Aug 18 10:34:32] DEBUG[15321] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '72b609e44febd02a23174a4d311879ff@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK73a7e9a8;received=178.62.121.41 From: ;tag=as67bc1c22 To: ;tag=as22df0306 Call-ID: 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[15322] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[15322] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Initializing initreq for method INVITE - callid 7530856d32712b46529332693455f8c4@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116831@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 3 [ 52]: From: ;tag=as655f7a70 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 6 [ 60]: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15310] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15322] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116831@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 Max-Forwards: 70 From: ;tag=as655f7a70 To: Contact: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2053228906 2053228906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19488 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Aug 18 10:34:32] DEBUG[15322] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15048] bridge_channel.c: Setting 0x7f0ca803dbf0(SIP/zvonobot-00000044) state from:0 to:1 [Aug 18 10:34:32] DEBUG[15321] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #152 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #152)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #22 (2) BYE - 8 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #22)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: BYE sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK204acf50 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41:5060", nonce="0de4ceb7", response="5f946298dc53c0cdee5695c18ecb459f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116836@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 Max-Forwards: 70 From: ;tag=as2d4671b8 To: Contact: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1289209748 1289209748 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #27)) [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE add candidate: 159.65.48.104:11192, 2130706431 [Aug 18 10:34:32] DEBUG[15048] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: pulling 0x7f0ca803dbf0(SIP/zvonobot-00000044) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116848@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0de5cf9c Max-Forwards: 70 From: ;tag=as517a3e83 To: Contact: Call-ID: 5501b2a8317acf721cd81f872ff0668c@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 608461512 608461512 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11040 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[15048] bridge_channel.c: Channel SIP/zvonobot-00000044 left 'simple_bridge' stasis-bridge <413b28bc-b121-462c-8ad3-b989ef736d5a> [Aug 18 10:34:32] DEBUG[15048] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: 0x7f0ca803dbf0(SIP/zvonobot-00000044) is leaving simple_bridge technology [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE add candidate: 10.131.0.10:13638, 2130706431 [Aug 18 10:34:32] VERBOSE[15322] dial.c: Called zvonobot/79821116831 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #117 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15310] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #117)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116829@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 Max-Forwards: 70 From: ;tag=as5c283596 To: Contact: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2033823795 2033823795 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000044 - start 1629282834.842178 answer 1629282867.366729 end 1629282872.536141 dur 37.693 bill 5.169 dispo ANSWERED [Aug 18 10:34:32] DEBUG[15048] bridge_native_rtp.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] DEBUG[15048] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15048] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15048] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15048] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15048] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a is already using the new technology. [Aug 18 10:34:32] DEBUG[15310] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[15055] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: Allocating new SIP dialog for 7de8b66b20ddc504657299bf115ee9f6@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15055] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15048] stasis/control.c: 213028, 413b28bc-b121-462c-8ad3-b989ef736d5a: Channel was departed from bridge [Aug 18 10:34:32] DEBUG[15315] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[15106] app.c: One waitfor failed, trying another [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE add candidate: 10.131.0.10:11192, 2130706431 [Aug 18 10:34:32] DEBUG[15310] rtp_engine.c: RTP instance '0x7f0c900ac120' is setup and ready to go [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) ICE stopped [Aug 18 10:34:32] DEBUG[15310] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15310] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15310] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15310] res_rtp_asterisk.c: (0x7f0c900ac120) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15310] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15310] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15310] chan_sip.c: SIP call-id changed from '2531f6200f4aca5350525ec934573a9c@127.0.1.1:5060' to '7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15310] stasis.c: Creating topic. name: channel:213217, detail: [Aug 18 10:34:32] DEBUG[15310] stasis.c: Topic 'channel:213217': 0x7f0c900abf30 created [Aug 18 10:34:32] DEBUG[15310] stasis.c: Creating topic. name: cache:766/channel:213217, detail: [Aug 18 10:34:32] DEBUG[15310] stasis.c: Topic 'cache:766/channel:213217': 0x7f0c90050e20 created [Aug 18 10:34:32] DEBUG[15048] stasis/app.c: bridge '413b28bc-b121-462c-8ad3-b989ef736d5a': is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[14963] channel.c: Channel 0x7f0cac02d090 'Recorder/ARI-00000043;1' destroying [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2baf6b12 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116862@178.62.121.41", nonce="0b319825", response="6a20eac9692b8e193e543b5bdc77a717" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #113 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #113)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116850@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69e70797 Max-Forwards: 70 From: ;tag=as09a15a28 To: Contact: Call-ID: 6ba2731c352a981429345b965d229ce4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 662155204 662155204 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17618 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #25 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #25)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116833@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d Max-Forwards: 70 From: ;tag=as235fcd7f To: Contact: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1532262111 1532262111 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14362 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15051] channel.c: Channel 0x7f0c2c1003c0 'Recorder/ARI-00000052;2' allocated [Aug 18 10:34:32] DEBUG[15051] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:32] DEBUG[14917] channel.c: Channel 0x7f0c74046e10 'Announcer/ARI-00000042;2' destroying [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Session timer stopped: 4 - 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757;received=159.65.48.104 From: ;tag=as02885f54 To: ;tag=as47561d08 Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 716099691 716099691 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 16646 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0c4c2757;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as02885f54 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as47561d08 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 264 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 1 [ 47]: o=root 716099691 716099691 IN IP4 178.62.121.41 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 16646 RTP/AVP 0 8 101 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 10507dcf059680b46ad884550335c862@159.65.48.104:5060 (Checking To) --From tag as02885f54 --To-tag as47561d08 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 10507dcf059680b46ad884550335c862@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 825 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348;received=159.65.48.104 From: ;tag=as67ede665 To: ;tag=as495e7ce0 Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a032c56" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK11e3f348;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as67ede665 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as495e7ce0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a032c56" [Aug 18 10:34:32] DEBUG[15048] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:32] DEBUG[13426] stasis/control.c: 213028: Channel departing bridge [Aug 18 10:34:32] DEBUG[13426] bridge.c: Waiting for 0x7f0ca803dbf0(SIP/zvonobot-00000044) bridge thread to die. [Aug 18 10:34:32] DEBUG[15323] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: 0x7f0c2c0536d0(Recorder/ARI-00000052;2) is joining [Aug 18 10:34:32] DEBUG[15319] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca40085b0' [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15315] chan_sip.c: SIP call-id changed from '7acb35612eb2724741b79a8a4c620ad7@127.0.1.1:5060' to '509f8e0e3acc6f0a7d623d30256c2b1f@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15315] stasis.c: Creating topic. name: channel:213218, detail: [Aug 18 10:34:32] DEBUG[15315] stasis.c: Topic 'channel:213218': 0x7f0ca811b090 created [Aug 18 10:34:32] DEBUG[15315] stasis.c: Creating topic. name: cache:767/channel:213218, detail: [Aug 18 10:34:32] DEBUG[15321] rtp_engine.c: RTP instance '0x7f0cac01e130' is setup and ready to go [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) ICE stopped [Aug 18 10:34:32] DEBUG[15321] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15321] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[13426] stasis/app.c: channel '213028': is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15321] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15052] channel.c: Channel 0x7f0c2815a320 'Announcer/ARI-0000005a;2' allocated [Aug 18 10:34:32] DEBUG[15064] stasis.c: Creating topic. name: channel:1629282872.673, detail: [Aug 18 10:34:32] DEBUG[14963] stasis.c: Destroying topic. name: cache:498/channel:1629282859.434, detail: [Aug 18 10:34:32] DEBUG[14963] stasis.c: Topic 'cache:498/channel:1629282859.434': 0x7f0cac06c570 destroyed [Aug 18 10:34:32] DEBUG[14963] stasis.c: Destroying topic. name: channel:1629282859.434, detail: [Aug 18 10:34:32] DEBUG[14963] stasis.c: Topic 'channel:1629282859.434': 0x7f0cac00bd00 destroyed [Aug 18 10:34:32] DEBUG[15064] stasis.c: Topic 'channel:1629282872.673': 0x7f0c7802c930 created [Aug 18 10:34:32] DEBUG[15064] stasis.c: Creating topic. name: cache:768/channel:1629282872.673, detail: [Aug 18 10:34:32] DEBUG[15064] stasis.c: Topic 'cache:768/channel:1629282872.673': 0x7f0c7802b570 created [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) RTP allocated port 18248 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] DEBUG[15321] res_rtp_asterisk.c: (0x7f0cac01e130) RTCP setup on RTP instance [Aug 18 10:34:32] DEBUG[15052] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:32] DEBUG[15052] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000005a;1' [Aug 18 10:34:32] DEBUG[13426] channel.c: Channel 0x7f0c3c04f300 'SIP/zvonobot-00000044' hanging up. Refs: 2 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 465 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 465 [Aug 18 10:34:32] DEBUG[15315] stasis.c: Topic 'cache:767/channel:213218': 0x7f0ca810af10 created [Aug 18 10:34:32] VERBOSE[15321] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15321] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15323] bridge_channel.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: pushing 0x7f0c2c0536d0(Recorder/ARI-00000052;2) [Aug 18 10:34:32] DEBUG[15321] chan_sip.c: SIP call-id changed from '73829489627e34e909638d6027fbd19c@127.0.1.1:5060' to '32d69425685fae543fe99f5e243b59f6@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE creating session 0.0.0.0:18248 (18248) [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE create [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE add system candidates [Aug 18 10:34:32] DEBUG[15319] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15319] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE add candidate: 159.65.48.104:18248, 2130706431 [Aug 18 10:34:32] DEBUG[15319] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15319] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE add candidate: 10.131.0.10:18248, 2130706431 [Aug 18 10:34:32] DEBUG[15319] rtp_engine.c: RTP instance '0x7f0ca40085b0' is setup and ready to go [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) ICE stopped [Aug 18 10:34:32] DEBUG[15319] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15319] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15319] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15319] res_rtp_asterisk.c: (0x7f0ca40085b0) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15319] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15319] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15319] chan_sip.c: SIP call-id changed from '7de8b66b20ddc504657299bf115ee9f6@127.0.1.1:5060' to '359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15319] stasis.c: Creating topic. name: channel:213222, detail: [Aug 18 10:34:32] DEBUG[15319] stasis.c: Topic 'channel:213222': 0x7f0ca4061ea0 created [Aug 18 10:34:32] DEBUG[15319] stasis.c: Creating topic. name: cache:769/channel:213222, detail: [Aug 18 10:34:32] DEBUG[15319] stasis.c: Topic 'cache:769/channel:213222': 0x7f0ca4064550 created [Aug 18 10:34:32] DEBUG[15323] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:32] DEBUG[15324] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c280f4d10(Announcer/ARI-0000005a;2) is joining [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 (Checking To) --From tag as67ede665 --To-tag as495e7ce0 [Aug 18 10:34:32] VERBOSE[15323] bridge_channel.c: Channel Recorder/ARI-00000052;2 joined 'simple_bridge' stasis-bridge <413b28bc-b121-462c-8ad3-b989ef736d5a> [Aug 18 10:34:32] DEBUG[15324] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pushing 0x7f0c280f4d10(Announcer/ARI-0000005a;2) [Aug 18 10:34:32] DEBUG[14917] stasis.c: Destroying topic. name: cache:548/channel:1629282862.476, detail: [Aug 18 10:34:32] DEBUG[14917] stasis.c: Topic 'cache:548/channel:1629282862.476': 0x7f0c74053dd0 destroyed [Aug 18 10:34:32] DEBUG[14917] stasis.c: Destroying topic. name: channel:1629282862.476, detail: [Aug 18 10:34:32] DEBUG[14917] stasis.c: Topic 'channel:1629282862.476': 0x7f0c74071210 destroyed [Aug 18 10:34:32] DEBUG[14882] channel.c: Channel 0x7f0c940b0650 'Announcer/ARI-0000003f;2' destroying [Aug 18 10:34:32] DEBUG[15323] bridge_native_rtp.c: Bridge '413b28bc-b121-462c-8ad3-b989ef736d5a' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] DEBUG[15323] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15323] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15323] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15323] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15323] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a is already using the new technology. [Aug 18 10:34:32] DEBUG[15323] bridge.c: Bridge 413b28bc-b121-462c-8ad3-b989ef736d5a: 0x7f0c2c0536d0(Recorder/ARI-00000052;2) is joining simple_bridge technology [Aug 18 10:34:32] DEBUG[15051] res_stasis_recording.c: 1629282867.573: Sending record(213028_guFLOBpoQdRZKIhmRxjdjWZQKKcAqmNk.wav) command [Aug 18 10:34:32] DEBUG[15051] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:32] DEBUG[15051] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15327] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15327] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:32] DEBUG[15321] stasis.c: Creating topic. name: channel:213223, detail: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2af2115663f68a66381dde6b32251ca1@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15197] channel.c: Channel 0x7f0c20037750 'SIP/zvonobot-000000f6' allocated [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15197] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[15321] stasis.c: Topic 'channel:213223': 0x7f0cac06c570 created [Aug 18 10:34:32] DEBUG[15327] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15327] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15327] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15327] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15326] app.c: play_and_record: , /var/spool/asterisk/recording/213028_guFLOBpoQdRZKIhmRxjdjWZQKKcAqmNk, 'wav' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Finding handler for bridges [Aug 18 10:34:32] DEBUG[15324] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Finding handler for bridges [Aug 18 10:34:32] DEBUG[15326] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:32] DEBUG[14882] stasis.c: Destroying topic. name: cache:531/channel:1629282860.461, detail: [Aug 18 10:34:32] DEBUG[14882] stasis.c: Topic 'cache:531/channel:1629282860.461': 0x7f0c94068690 destroyed [Aug 18 10:34:32] DEBUG[14882] stasis.c: Destroying topic. name: channel:1629282860.461, detail: [Aug 18 10:34:32] DEBUG[14882] stasis.c: Topic 'channel:1629282860.461': 0x7f0c940aea90 destroyed [Aug 18 10:34:32] DEBUG[15197] res_stasis.c: calls_0: Subscribing to 213208 [Aug 18 10:34:32] DEBUG[15197] stasis/app.c: Channel '213208' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:32] DEBUG[15321] stasis.c: Creating topic. name: cache:770/channel:213223, detail: [Aug 18 10:34:32] VERBOSE[15326] app.c: x=0, open writing: /var/spool/asterisk/recording/213028_guFLOBpoQdRZKIhmRxjdjWZQKKcAqmNk format: wav, 0x7f0c08060030 [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:32] DEBUG[15321] stasis.c: Topic 'cache:770/channel:213223': 0x7f0cac002d30 created [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:32] DEBUG[15197] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[15197] http.c: HTTP closing session. Top level [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706901 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #142 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #142)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090437 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116831@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 Max-Forwards: 70 From: ;tag=as655f7a70 To: Contact: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2053228906 2053228906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19488 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #86 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #86)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876615 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573;received=159.65.48.104 From: ;tag=as05f1fb09 To: ;tag=as5f4f7e61 Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a2a9519" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20ed6573;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as05f1fb09 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5f4f7e61 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a2a9519" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 (Checking To) --From tag as05f1fb09 --To-tag as5f4f7e61 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 41cffb51539db62640feb00322cb29ef@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #25 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #25)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116833@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d Max-Forwards: 70 From: ;tag=as235fcd7f To: Contact: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1532262111 1532262111 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14362 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Destroying SIP dialog 189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '189f186a0344dd2e0e31acb72c88017e@159.65.48.104:5060' Method: BYE [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS stop [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c8c00f7e0) ICE RTP transport deallocating [Aug 18 10:34:32] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c8c00f7e0' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Destroying SIP dialog 17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '17764d9f28db06e82a6712eb07f409e7@159.65.48.104:5060' Method: BYE [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS stop [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c70049e60) ICE RTP transport deallocating [Aug 18 10:34:32] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c70049e60' [Aug 18 10:34:32] DEBUG[14858] chan_sip.c: Hangup call SIP/zvonobot-000000c3, SIP callid 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[15324] bridge_channel.c: Channel Announcer/ARI-0000005a;2 joined 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:32] DEBUG[15210] channel.c: Channel 0x7f0c840700c0 'SIP/zvonobot-000000f7' allocated [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: Allocating new SIP dialog for 66510cbd0350170d34e65403298f3285@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15320] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb0023990' [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) RTP allocated port 15326 [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE creating session 0.0.0.0:15326 (15326) [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE create [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE add system candidates [Aug 18 10:34:32] DEBUG[15320] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15320] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE add candidate: 159.65.48.104:15326, 2130706431 [Aug 18 10:34:32] DEBUG[15320] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15320] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE add candidate: 10.131.0.10:15326, 2130706431 [Aug 18 10:34:32] DEBUG[15320] rtp_engine.c: RTP instance '0x7f0cb0023990' is setup and ready to go [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) ICE stopped [Aug 18 10:34:32] DEBUG[15320] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15320] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15320] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[15320] res_rtp_asterisk.c: (0x7f0cb0023990) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15320] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15320] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Outgoing Call for 79821116832 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15210] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[14858] res_rtp_asterisk.c: (0x7f0c840983e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14858] res_rtp_asterisk.c: (0x7f0c840983e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14858] channel.c: Channel 0x7f0c8411eca0 'SIP/zvonobot-000000c3' destroying [Aug 18 10:34:32] DEBUG[15327] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213160': is 0 interested in calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213160' unsubscribed from calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: cache:525/channel:213160, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'cache:525/channel:213160': 0x7f0c8406c130 destroyed [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: channel:213160, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'channel:213160': 0x7f0c8406c080 destroyed [Aug 18 10:34:32] DEBUG[15327] stasis.c: Creating topic. name: bridge:0e24edfd-8541-4e8b-acb7-53cd7c3058be, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282872.676, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.676': 0x7f0c3003b930 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: cache:771/channel:1629282872.676, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:771/channel:1629282872.676': 0x7f0c300413e0 created [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15328] chan_sip.c: Audio is at 16686 [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: cache:771/channel:1629282872.676, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:771/channel:1629282872.676': 0x7f0c300413e0 destroyed [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282872.676, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.676': 0x7f0c3003b930 destroyed [Aug 18 10:34:32] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:24', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000c3', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213160', '')] [Aug 18 10:34:32] VERBOSE[15328] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15328] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] DEBUG[15327] stasis.c: Topic 'bridge:0e24edfd-8541-4e8b-acb7-53cd7c3058be': 0x7f0c1c066ba0 created [Aug 18 10:34:32] DEBUG[15327] stasis.c: Creating topic. name: cache:772/bridge:0e24edfd-8541-4e8b-acb7-53cd7c3058be, detail: [Aug 18 10:34:32] DEBUG[15324] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] VERBOSE[15328] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15327] stasis.c: Topic 'cache:772/bridge:0e24edfd-8541-4e8b-acb7-53cd7c3058be': 0x7f0c1c00ee40 created [Aug 18 10:34:32] DEBUG[15210] res_stasis.c: calls_0: Subscribing to 213212 [Aug 18 10:34:32] DEBUG[15210] stasis/app.c: Channel '213212' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15327] bridge_native_rtp.c: Bridge '0e24edfd-8541-4e8b-acb7-53cd7c3058be' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] DEBUG[15327] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15324] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15210] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[15324] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15327] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:32] DEBUG[15327] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15327] bridge.c: Bridge 0e24edfd-8541-4e8b-acb7-53cd7c3058be: calling simple_bridge technology constructor [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Outgoing Call for 79821116828 [Aug 18 10:34:32] DEBUG[15327] bridge.c: Bridge 0e24edfd-8541-4e8b-acb7-53cd7c3058be: calling simple_bridge technology start [Aug 18 10:34:32] DEBUG[15210] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[15324] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15324] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[15324] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15329] chan_sip.c: Audio is at 15460 [Aug 18 10:34:32] VERBOSE[15329] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15329] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] DEBUG[15327] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[15324] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c280f4d10(Announcer/ARI-0000005a;2) is joining simple_bridge technology [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: Allocating new SIP dialog for 3232d9372c4c07b16219b6de41bf3738@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:32] DEBUG[15317] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9808ec70' [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) RTP allocated port 12594 [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE creating session 0.0.0.0:12594 (12594) [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE create [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE add system candidates [Aug 18 10:34:32] DEBUG[15317] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:32] DEBUG[15317] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE add candidate: 159.65.48.104:12594, 2130706431 [Aug 18 10:34:32] DEBUG[15317] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:32] DEBUG[15317] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE add candidate: 10.131.0.10:12594, 2130706431 [Aug 18 10:34:32] DEBUG[15317] rtp_engine.c: RTP instance '0x7f0c9808ec70' is setup and ready to go [Aug 18 10:34:32] DEBUG[15327] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15330] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15330] http.c: HTTP Request URI is /ari/channels/213028/snoop?app=calls_0&spy=in [Aug 18 10:34:32] DEBUG[15330] http.c: match request [ari/channels/213028/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15320] chan_sip.c: SIP call-id changed from '66510cbd0350170d34e65403298f3285@127.0.1.1:5060' to '7e9a053f122d7f9206b4bd144cc6c1b5@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15330] http.c: match request [ari/channels/213028/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15330] http.c: match request [ari/channels/213028/snoop] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15330] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15320] stasis.c: Creating topic. name: channel:213220, detail: [Aug 18 10:34:32] DEBUG[15052] res_stasis_playback.c: 1629282870.621: Sending play(sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4) command [Aug 18 10:34:32] DEBUG[15320] stasis.c: Topic 'channel:213220': 0x7f0cb000e690 created [Aug 18 10:34:32] DEBUG[15320] stasis.c: Creating topic. name: cache:773/channel:213220, detail: [Aug 18 10:34:32] DEBUG[15320] stasis.c: Topic 'cache:773/channel:213220': 0x7f0cb0050700 created [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Initializing initreq for method INVITE - callid 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116832@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 3 [ 52]: From: ;tag=as7025f96b [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 6 [ 60]: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15328] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116832@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 Max-Forwards: 70 From: ;tag=as7025f96b To: Contact: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 58040179 58040179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16686 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39 [Aug 18 10:34:32] DEBUG[15328] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888;received=159.65.48.104 From: ;tag=as4bc9e76f To: ;tag=as5fdc366e Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="328f7607" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4bc9e76f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5fdc366e [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="328f7607" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 (Checking To) --From tag as4bc9e76f --To-tag as5fdc366e [Aug 18 10:34:32] DEBUG[15052] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) ICE stopped [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Finding handler for channels/213028/snoop [Aug 18 10:34:32] DEBUG[15317] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:32] DEBUG[15317] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:32] DEBUG[15052] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:32] VERBOSE[15328] dial.c: Called zvonobot/79821116832 [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Finding handler for channels [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #123 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Stopping retransmission on '0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888 Max-Forwards: 70 From: ;tag=as4bc9e76f To: ;tag=as5fdc366e Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Audio is at 16814 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] VERBOSE[15329] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Initializing initreq for method INVITE - callid 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116828@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 3 [ 52]: From: ;tag=as7b799277 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 6 [ 60]: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15329] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116828@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df Max-Forwards: 70 From: ;tag=as7b799277 To: Contact: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 670818967 670818967 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #24 [Aug 18 10:34:32] DEBUG[15329] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15332] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Finding handler for 213028 [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channels create: Didn't match 213028 [Aug 18 10:34:32] DEBUG[15317] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 385 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 385 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channels externalMedia: Didn't match 213028 [Aug 18 10:34:32] DEBUG[15330] res_ari.c: No explicit handler found for 213028. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Finding handler for snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:32] DEBUG[15330] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:32] VERBOSE[15329] dial.c: Called zvonobot/79821116828 [Aug 18 10:34:32] DEBUG[15332] http.c: HTTP Request URI is /ari/playbacks/67aa63c7-114a-4eeb-a87c-a179a3d3f2fe [Aug 18 10:34:32] DEBUG[15332] http.c: match request [ari/playbacks/67aa63c7-114a-4eeb-a87c-a179a3d3f2fe] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15332] http.c: match request [ari/playbacks/67aa63c7-114a-4eeb-a87c-a179a3d3f2fe] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15332] http.c: match request [ari/playbacks/67aa63c7-114a-4eeb-a87c-a179a3d3f2fe] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15331] channel.c: Channel Announcer/ARI-0000005a;1 setting write format path: gsm -> slin [Aug 18 10:34:32] DEBUG[15332] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Finding handler for playbacks/67aa63c7-114a-4eeb-a87c-a179a3d3f2fe [Aug 18 10:34:32] DEBUG[15317] res_rtp_asterisk.c: (0x7f0c9808ec70) RTCP setup on RTP instance [Aug 18 10:34:32] VERBOSE[15317] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:32] DEBUG[15317] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Finding handler for playbacks [Aug 18 10:34:32] DEBUG[15317] chan_sip.c: SIP call-id changed from '3232d9372c4c07b16219b6de41bf3738@127.0.1.1:5060' to '2aede3cb384162731ba8d58b471d2348@159.65.48.104:5060' [Aug 18 10:34:32] DEBUG[15317] stasis.c: Creating topic. name: channel:213221, detail: [Aug 18 10:34:32] DEBUG[15317] stasis.c: Topic 'channel:213221': 0x7f0c98007090 created [Aug 18 10:34:32] DEBUG[15317] stasis.c: Creating topic. name: cache:774/channel:213221, detail: [Aug 18 10:34:32] DEBUG[15317] stasis.c: Topic 'cache:774/channel:213221': 0x7f0c980363c0 created [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari endpoints: Didn't match playbacks [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari events: Didn't match playbacks [Aug 18 10:34:32] DEBUG[15331] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116865@178.62.121.41", nonce="328f7607", response="03e2ed289df65658183229073b55646c" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Aug 18 10:34:32] VERBOSE[15331] file.c: Playing '/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4.gsm' (language 'en') [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari applications: Didn't match playbacks [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari recordings: Didn't match playbacks [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari channels: Didn't match playbacks [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (4) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #153 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #153)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116853@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b4e9151 Max-Forwards: 70 From: ;tag=as6ae39521 To: Contact: Call-ID: 5d12ba41133c6fc357251e5b37ca5649@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 155651249 155651249 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12262 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Destroying SIP dialog 143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '143f4ea6098e4b9c3353269e036c9795@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c840983e0) DTLS stop [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c840983e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c840983e0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c840983e0) ICE RTP transport deallocating [Aug 18 10:34:32] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c840983e0' [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #155 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #155)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116855@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK49840aaf Max-Forwards: 70 From: ;tag=as76c090e3 To: Contact: Call-ID: 21405c0201d61407119338763dc16673@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1812849690 1812849690 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14186 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #157 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #157)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116852@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK50876d25 Max-Forwards: 70 From: ;tag=as6137e79e To: Contact: Call-ID: 7a24ed7331ac755460e7b39d39300ee8@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 528097032 528097032 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11578 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #119 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #119)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116834@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc Max-Forwards: 70 From: ;tag=as2f225ac7 To: Contact: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637714792 637714792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10426 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116835@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 Max-Forwards: 70 From: ;tag=as2e8baab5 To: Contact: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1705768465 1705768465 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17162 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96;received=159.65.48.104 From: ;tag=as14ba6e32 To: ;tag=as60c8efa2 Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7aa13769" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK79c38b96;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as14ba6e32 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as60c8efa2 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7aa13769" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 (Checking To) --From tag as14ba6e32 --To-tag as60c8efa2 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 279f3b83786d99fd34f973337c017e8c@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[15211] channel.c: Channel 0x7f0c800774a0 'SIP/zvonobot-000000f8' allocated [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] DEBUG[15211] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari bridges: Didn't match playbacks [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking ari playbacks: Explicit match with playbacks [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Finding handler for 67aa63c7-114a-4eeb-a87c-a179a3d3f2fe [Aug 18 10:34:32] DEBUG[15211] res_stasis.c: calls_0: Subscribing to 213213 [Aug 18 10:34:32] DEBUG[15211] stasis/app.c: Channel '213213' is 1 interested in calls_0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6;received=159.65.48.104 From: ;tag=as0cd290ec To: ;tag=as01d9622d Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5ae8ae" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[15211] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51d747b6;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0cd290ec [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as01d9622d [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5ae8ae" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 (Checking To) --From tag as0cd290ec --To-tag as01d9622d [Aug 18 10:34:32] DEBUG[15211] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15332] res_ari.c: Checking playbacks playbackId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15332] res_ari.c: No explicit handler found for 67aa63c7-114a-4eeb-a87c-a179a3d3f2fe. Using wildcard playbackId. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3ddae3401cf3b7bf1dcc20c40ea8ef27@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[15332] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:32] DEBUG[15332] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[15331] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Outgoing Call for 79821116827 [Aug 18 10:34:32] DEBUG[15336] http.c: HTTP opening session. Top level [Aug 18 10:34:32] DEBUG[15336] http.c: HTTP Request URI is /ari/channels/212967 [Aug 18 10:34:32] DEBUG[15336] http.c: match request [ari/channels/212967] with handler [httpstatus] len 10 [Aug 18 10:34:32] DEBUG[15336] http.c: match request [ari/channels/212967] with handler [phoneprov] len 9 [Aug 18 10:34:32] DEBUG[15336] http.c: match request [ari/channels/212967] with handler [ari] len 3 [Aug 18 10:34:32] DEBUG[15336] http.c: Match made with [ari] [Aug 18 10:34:32] DEBUG[15331] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:32] DEBUG[15331] channel.c: Channel Announcer/ARI-0000005a;1 setting write format path: slin -> slin [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 From: ;tag=as697b28a1 To: ;tag=as7ad66467 Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK32284001;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as697b28a1 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7ad66467 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390b6417" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 (Checking To) --From tag as697b28a1 --To-tag as7ad66467 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 00847b4b4695b1953a84dacb3b91817a@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Finding handler for channels/212967 [Aug 18 10:34:32] NOTICE[15331] res_stasis_playback.c: 1629282870.621: Playback stopped for sound:/var/www/storage/audio/d4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Finding handler for channels [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 From: ;tag=as1c2a52a2 To: ;tag=as025c937f Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK587a4abe;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as1c2a52a2 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as025c937f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46c0a6b9" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 (Checking To) --From tag as1c2a52a2 --To-tag as025c937f [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 0fc30825773f0fd33ee04e567600aa31@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (2) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116831@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 Max-Forwards: 70 From: ;tag=as655f7a70 To: Contact: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2053228906 2053228906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19488 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 383 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 383 [Aug 18 10:34:32] DEBUG[15331] channel.c: Channel 0x7f0c280b5380 'Announcer/ARI-0000005a;1' hanging up. Refs: 2 [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Finding handler for 212967 [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking channels create: Didn't match 212967 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f;received=159.65.48.104 From: ;tag=as0228d9c2 To: ;tag=as5e469a68 Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cb8ffaa" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:32] DEBUG[15336] res_ari.c: Checking channels externalMedia: Didn't match 212967 [Aug 18 10:34:32] DEBUG[15336] res_ari.c: No explicit handler found for 212967. Using wildcard channelId. [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0228d9c2 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5e469a68 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cb8ffaa" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 (Checking To) --From tag as0228d9c2 --To-tag as5e469a68 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #121 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Stopping retransmission on '447a50ba7810af590f60b91a27726642@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK220c4f3f Max-Forwards: 70 From: ;tag=as0228d9c2 To: ;tag=as5e469a68 Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Audio is at 19846 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b84ac14 Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116859@178.62.121.41", nonce="4cb8ffaa", response="34d7073f228c83199f1d8a15ca6aeca9" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23;received=159.65.48.104 From: ;tag=as17ad889a To: ;tag=as778331a1 Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09aacdf1" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as17ad889a [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as778331a1 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09aacdf1" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15335] chan_sip.c: Audio is at 15356 [Aug 18 10:34:32] VERBOSE[15335] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[15335] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 (Checking To) --From tag as17ad889a --To-tag as778331a1 [Aug 18 10:34:32] VERBOSE[15335] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Initializing initreq for method INVITE - callid 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116827@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 3 [ 52]: From: ;tag=as568913af [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 6 [ 60]: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] VERBOSE[15335] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116827@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d Max-Forwards: 70 From: ;tag=as568913af To: Contact: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 962154366 962154366 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #95 [Aug 18 10:34:32] DEBUG[15335] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Stopping retransmission on '2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK62533b23 Max-Forwards: 70 From: ;tag=as17ad889a To: ;tag=as778331a1 Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] VERBOSE[15335] dial.c: Called zvonobot/79821116827 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Audio is at 13594 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26886e91 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116860@178.62.121.41", nonce="09aacdf1", response="3c5bdea99ab5d5c94251dcfc28700249" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890281 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #112 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #22 (3) BYE - 8 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #22)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: BYE sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK204acf50 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41:5060", nonce="0de4ceb7", response="5f946298dc53c0cdee5695c18ecb459f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15205] channel.c: Channel 0x7f0c74079320 'SIP/zvonobot-000000f9' allocated [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc;received=159.65.48.104 From: ;tag=as6b79f1a3 To: ;tag=as0e105aab Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ecc103f" Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc;received=159.65.48.104 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6b79f1a3 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as0e105aab [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ecc103f" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 (Checking To) --From tag as6b79f1a3 --To-tag as0e105aab [Aug 18 10:34:32] DEBUG[15205] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:32] DEBUG[15205] res_stasis.c: calls_0: Subscribing to 213210 [Aug 18 10:34:32] DEBUG[15205] stasis/app.c: Channel '213210' is 1 interested in calls_0 [Aug 18 10:34:32] DEBUG[15205] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Stopping retransmission on '08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Outgoing Call for 79821116830 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22b1d9dc Max-Forwards: 70 From: ;tag=as6b79f1a3 To: ;tag=as0e105aab Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Audio is at 12150 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06fea4c6 Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116856@178.62.121.41", nonce="4ecc103f", response="5984a684e9cf4d2ea38b7a835805bdb8" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029195 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #87 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15205] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #39 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #39)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116832@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 Max-Forwards: 70 From: ;tag=as7025f96b To: Contact: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 58040179 58040179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16686 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #117 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #117)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116829@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 Max-Forwards: 70 From: ;tag=as5c283596 To: Contact: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2033823795 2033823795 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (3) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2baf6b12 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116862@178.62.121.41", nonce="0b319825", response="6a20eac9692b8e193e543b5bdc77a717" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116828@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df Max-Forwards: 70 From: ;tag=as7b799277 To: Contact: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 670818967 670818967 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] WARNING[15105] app.c: No audio available on Recorder/ARI-00000041;1?? [Aug 18 10:34:32] DEBUG[15090] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:32] VERBOSE[15105] app.c: User hung up [Aug 18 10:34:32] DEBUG[15105] res_stasis_recording.c: 1629282858.423: Recording complete [Aug 18 10:34:32] DEBUG[15090] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[13271] chan_sip.c: Hangup call SIP/zvonobot-00000034, SIP callid 549831e20d284c8653320b2042dceffc@159.65.48.104:5060 [Aug 18 10:34:32] VERBOSE[15202] app.c: User hung up [Aug 18 10:34:32] DEBUG[15202] res_stasis_recording.c: 1629282863.500: Recording complete [Aug 18 10:34:32] DEBUG[15202] channel.c: Channel 0x7f0c08080020 'Recorder/ARI-0000004c;1' hanging up. Refs: 2 [Aug 18 10:34:32] DEBUG[15058] res_rtp_asterisk.c: (0x7f0c2003ba40) RTCP got report of 76 bytes from 178.62.121.41:16217 [Aug 18 10:34:32] DEBUG[15086] channel.c: Channel 0x7f0ca40622a0 'Announcer/ARI-00000054;2' allocated [Aug 18 10:34:32] DEBUG[15086] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:32] DEBUG[15086] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000054;1' [Aug 18 10:34:32] DEBUG[13271] res_rtp_asterisk.c: (0x7f0c20028ba0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[13271] res_rtp_asterisk.c: (0x7f0c20028ba0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[15174] channel.c: Channel 0x7f0c0807aab0 'Recorder/ARI-0000004c;2' destroying [Aug 18 10:34:32] VERBOSE[15058] res_rtp_asterisk.c: 0x7f0c20044340 -- Strict RTP learning complete - Locking on source address 178.62.121.41:16216 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:32] DEBUG[13271] channel.c: Channel 0x7f0c20031440 'SIP/zvonobot-00000034' destroying [Aug 18 10:34:32] DEBUG[15105] channel.c: Channel 0x7f0c30037010 'Recorder/ARI-00000041;1' hanging up. Refs: 2 [Aug 18 10:34:32] DEBUG[14889] channel.c: Channel 0x7f0c38008fe0 'Snoop/213002-0000001c' allocated [Aug 18 10:34:32] DEBUG[14889] channel.c: Channel 0x7f0c38008fe0 'Snoop/213002-0000001c' hanging up. Refs: 3 [Aug 18 10:34:32] DEBUG[14889] autoservice.c: Thread is a user interface, not removing channel Snoop/213002-0000001c from autoservice [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:32] VERBOSE[15344] chan_sip.c: Audio is at 16394 [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282872.679, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.679': 0x7f0c3003bd10 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: cache:775/channel:1629282872.679, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:775/channel:1629282872.679': 0x7f0c300b9e40 created [Aug 18 10:34:32] DEBUG[15346] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0ca4062070(Announcer/ARI-00000054;2) is joining [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213016': is 0 interested in calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis/app.c: channel '213016' unsubscribed from calls_0 [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: cache:88/channel:213016, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'cache:88/channel:213016': 0x7f0c200262c0 destroyed [Aug 18 10:34:32] DEBUG[20620] stasis.c: Destroying topic. name: channel:213016, detail: [Aug 18 10:34:32] DEBUG[20620] stasis.c: Topic 'channel:213016': 0x7f0c2002ec20 destroyed [Aug 18 10:34:32] VERBOSE[15344] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: cache:775/channel:1629282872.679, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:775/channel:1629282872.679': 0x7f0c300b9e40 destroyed [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282872.679, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.679': 0x7f0c3003bd10 destroyed [Aug 18 10:34:32] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000034', '', 'Stasis', 'calls_0', 38, 8, 'ANSWERED', 3, '', '213016', '')] [Aug 18 10:34:32] DEBUG[15174] stasis.c: Destroying topic. name: cache:640/channel:1629282867.557, detail: [Aug 18 10:34:32] DEBUG[15174] stasis.c: Topic 'cache:640/channel:1629282867.557': 0x7f0c0802cbd0 destroyed [Aug 18 10:34:32] DEBUG[15174] stasis.c: Destroying topic. name: channel:1629282867.557, detail: [Aug 18 10:34:32] DEBUG[15174] stasis.c: Topic 'channel:1629282867.557': 0x7f0c08074300 destroyed [Aug 18 10:34:32] VERBOSE[15344] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:32] DEBUG[15346] bridge_channel.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: pushing 0x7f0ca4062070(Announcer/ARI-00000054;2) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK01c72337 Max-Forwards: 70 From: ;tag=as58d484f0 To: ;tag=as08e169d8 Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 102 BYE User-Agent: Asterisk PBX 16.18.0 Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="560ef8b8", response="178b0dac31627b6f983c3b5892ec7727" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 0 [ 46]: BYE sip:74950493843@159.65.48.104:5060 SIP/2.0 [Aug 18 10:34:32] DEBUG[15346] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:32] VERBOSE[15346] bridge_channel.c: Channel Announcer/ARI-00000054;2 joined 'simple_bridge' stasis-bridge <61075423-3ee2-4d60-8382-ee99e654a5be> [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK01c72337 [Aug 18 10:34:32] VERBOSE[15344] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:32] DEBUG[15346] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be'. Checking compatability for channels 'SIP/zvonobot-00000047' and 'Announcer/ARI-00000054;2' [Aug 18 10:34:32] DEBUG[15346] bridge_native_rtp.c: Bridge '61075423-3ee2-4d60-8382-ee99e654a5be' can not use native RTP bridge as channel 'SIP/zvonobot-00000047' has features which prevent it [Aug 18 10:34:32] DEBUG[15346] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:32] DEBUG[15346] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15346] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:32] DEBUG[15346] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:32] DEBUG[15346] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be is already using the new technology. [Aug 18 10:34:32] DEBUG[15346] bridge.c: Bridge 61075423-3ee2-4d60-8382-ee99e654a5be: 0x7f0ca4062070(Announcer/ARI-00000054;2) is joining simple_bridge technology [Aug 18 10:34:32] DEBUG[15346] channel.c: Channel Announcer/ARI-00000054;2 setting read format path: slin -> slin [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[15346] channel.c: Channel Announcer/ARI-00000054;2 setting write format path: slin -> slin [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 3 [ 52]: From: ;tag=as58d484f0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 4 [ 50]: To: ;tag=as08e169d8 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 5 [ 60]: Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 16.18.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 8 [168]: Proxy-Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:178.62.121.41", nonce="560ef8b8", response="178b0dac31627b6f983c3b5892ec7727" [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Initializing initreq for method INVITE - callid 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116830@178.62.121.41 SIP/2.0 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: --- (12 headers 0 lines) --- [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: = Looking for Call ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 (Checking From) --From tag as58d484f0 --To-tag as08e169d8 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 3 [ 52]: From: ;tag=as39b021ce [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Initializing initreq for method BYE - callid 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 6 [ 60]: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15086] res_stasis_playback.c: 1629282868.584: Sending play(sound:silence/2) command [Aug 18 10:34:32] DEBUG[20585] netsock2.c: Splitting '178.62.121.41:5060' into... [Aug 18 10:34:32] DEBUG[20585] netsock2.c: ...host '178.62.121.41' and port '5060'. [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Sending to 178.62.121.41:5060 (no NAT) [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:32] DEBUG[15086] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:32] DEBUG[15086] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:32 GMT [Aug 18 10:34:32] DEBUG[14447] channel.c: SIP/zvonobot-00000047: Dropping redundant connected line update "" <>. [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:32] DEBUG[15347] channel.c: Channel Announcer/ARI-00000054;1 setting write format path: gsm -> slin [Aug 18 10:34:32] DEBUG[14447] audiohook.c: Audiohook 0x7f0c08015b10 has stale audio in its factories. Flushing them both [Aug 18 10:34:32] DEBUG[14447] res_rtp_asterisk.c: (0x7f0ca804b700) RTP ooh, format changed from none to alaw [Aug 18 10:34:32] DEBUG[14447] res_rtp_asterisk.c: (0x7f0ca804b700) RTCP starting transmission [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:32] DEBUG[15347] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:32] VERBOSE[15347] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:32] VERBOSE[15344] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116830@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd Max-Forwards: 70 From: ;tag=as39b021ce To: Contact: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1100223183 1100223183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #89 [Aug 18 10:34:32] DEBUG[15344] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] VERBOSE[15344] dial.c: Called zvonobot/79821116830 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c1000e000) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Scheduling destruction of SIP dialog '261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060' in 6400 ms (Method: BYE) [Aug 18 10:34:32] DEBUG[14905] stasis.c: Creating topic. name: channel:1629282872.680, detail: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Received bye, issuing owner hangup [Aug 18 10:34:32] DEBUG[14905] stasis.c: Topic 'channel:1629282872.680': 0x7f0c9c0db590 created [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: <--- Transmitting (no NAT) to 178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 178.62.121.41:5060;branch=z9hG4bK01c72337;received=178.62.121.41 From: ;tag=as58d484f0 To: ;tag=as08e169d8 Call-ID: 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 CSeq: 102 BYE Server: Asterisk PBX 16.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Aug 18 10:34:32] DEBUG[14905] stasis.c: Creating topic. name: cache:776/channel:1629282872.680, detail: [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #4 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[14905] stasis.c: Topic 'cache:776/channel:1629282872.680': 0x7f0c9c058e20 created [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #4)) [Aug 18 10:34:32] DEBUG[14593] bridge_channel.c: Setting 0x7f0c08021e90(SIP/zvonobot-00000002) state from:0 to:1 [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116865@178.62.121.41", nonce="328f7607", response="03e2ed289df65658183229073b55646c" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[14734] channel.c: Channel 0x7f0c2419f2d0 'Announcer/ARI-00000038;1' destroying [Aug 18 10:34:32] DEBUG[14802] chan_sip.c: Hangup call SIP/zvonobot-000000bd, SIP callid 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[14802] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:32] DEBUG[14802] res_rtp_asterisk.c: (0x7f0c2c0c3c00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14802] res_rtp_asterisk.c: (0x7f0c2c0c3c00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14802] channel.c: Channel 0x7f0c2c0cc990 'SIP/zvonobot-000000bd' destroying [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:32] DEBUG[14660] chan_sip.c: Hangup call SIP/zvonobot-000000b0, SIP callid 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[14660] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14660] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14660] channel.c: Channel 0x7f0c2c09fc70 'SIP/zvonobot-000000b0' destroying [Aug 18 10:34:32] DEBUG[14895] chan_sip.c: Hangup call SIP/zvonobot-000000c6, SIP callid 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[15110] stasis.c: Creating topic. name: channel:1629282872.681, detail: [Aug 18 10:34:32] DEBUG[15110] stasis.c: Topic 'channel:1629282872.681': 0x7f0c380973e0 created [Aug 18 10:34:32] DEBUG[15110] stasis.c: Creating topic. name: cache:777/channel:1629282872.681, detail: [Aug 18 10:34:32] DEBUG[15110] stasis.c: Topic 'cache:777/channel:1629282872.681': 0x7f0c38097170 created [Aug 18 10:34:32] DEBUG[14895] chan_sip.c: Hanging up channel in state Down (not UP) [Aug 18 10:34:32] DEBUG[13896] channel.c: Channel 0x7f0c980222e0 'SIP/zvonobot-00000030' destroying [Aug 18 10:34:32] DEBUG[14895] res_rtp_asterisk.c: (0x7f0c3808cb20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14895] res_rtp_asterisk.c: (0x7f0c3808cb20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:32] DEBUG[14895] channel.c: Channel 0x7f0c3803b8c0 'SIP/zvonobot-000000c6' destroying [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282872.682, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.682': 0x7f0c3003bd10 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Creating topic. name: cache:778/channel:1629282872.682, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:778/channel:1629282872.682': 0x7f0c300413e0 created [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: cache:778/channel:1629282872.682, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'cache:778/channel:1629282872.682': 0x7f0c300413e0 destroyed [Aug 18 10:34:32] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282872.682, detail: [Aug 18 10:34:32] DEBUG[20545] stasis.c: Topic 'channel:1629282872.682': 0x7f0c3003bd10 destroyed [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b84ac14 Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116859@178.62.121.41", nonce="4cb8ffaa", response="34d7073f228c83199f1d8a15ca6aeca9" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #146 (5) INVITE - 5 [Aug 18 10:34:32] DEBUG[14734] stasis.c: Destroying topic. name: cache:419/channel:1629282854.363, detail: [Aug 18 10:34:32] DEBUG[14734] stasis.c: Topic 'cache:419/channel:1629282854.363': 0x7f0c241040e0 destroyed [Aug 18 10:34:32] DEBUG[14734] stasis.c: Destroying topic. name: channel:1629282854.363, detail: [Aug 18 10:34:32] DEBUG[14734] stasis.c: Topic 'channel:1629282854.363': 0x7f0c240761d0 destroyed [Aug 18 10:34:32] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:22', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000bd', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213147', '')] [Aug 18 10:34:32] DEBUG[14593] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: pulling 0x7f0c08021e90(SIP/zvonobot-00000002) [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #146)) [Aug 18 10:34:32] DEBUG[13896] channel.c: Channel 0x7f0c2c0b7210 'Snoop/213011-0000000f' destroying [Aug 18 10:34:32] VERBOSE[14593] bridge_channel.c: Channel SIP/zvonobot-00000002 left 'simple_bridge' stasis-bridge <79f92216-f8f4-49dd-85f1-f154853e1fd1> [Aug 18 10:34:32] DEBUG[13896] stasis.c: Destroying topic. name: cache:264/channel:1629282842.223, detail: [Aug 18 10:34:32] DEBUG[13896] stasis.c: Topic 'cache:264/channel:1629282842.223': 0x7f0c2c0ac570 destroyed [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116847@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3b439f80 Max-Forwards: 70 From: ;tag=as3fd361cb To: Contact: Call-ID: 5676735873320902534290d27970f7c7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 636710556 636710556 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13644 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[13896] stasis.c: Destroying topic. name: channel:1629282842.223, detail: [Aug 18 10:34:32] DEBUG[13896] stasis.c: Topic 'channel:1629282842.223': 0x7f0c2c07ec50 destroyed [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[14593] bridge_channel.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1: 0x7f0c08021e90(SIP/zvonobot-00000002) is leaving simple_bridge technology [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116827@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d Max-Forwards: 70 From: ;tag=as568913af To: Contact: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 962154366 962154366 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[15121] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:32] DEBUG[15121] http.c: HTTP closing session. Top level [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (1) INVITE - 5 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:32] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26886e91 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116860@178.62.121.41", nonce="09aacdf1", response="3c5bdea99ab5d5c94251dcfc28700249" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890281 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:32] DEBUG[20585] chan_sip.c: Session timer stopped: 9 - 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:32] DEBUG[14593] bridge_native_rtp.c: Bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1' can not use native RTP bridge as two channels are required [Aug 18 10:34:32] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:32] DEBUG[14593] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[14593] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[14593] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282872.683, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282872.683': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:779/channel:1629282872.683, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:779/channel:1629282872.683': 0x7f0c3003bd10 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:779/channel:1629282872.683, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:779/channel:1629282872.683': 0x7f0c3003bd10 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282872.683, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282872.683': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213147': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213147' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:500/channel:213147, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:500/channel:213147': 0x7f0c2c0ad5e0 destroyed [Aug 18 10:34:33] DEBUG[14593] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[14593] bridge.c: Bridge 79f92216-f8f4-49dd-85f1-f154853e1fd1 is already using the new technology. [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213147, detail: [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:25', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000c6', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213158', '')] [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213147': 0x7f0c2c0ce6e0 destroyed [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:33] DEBUG[13038] res_rtp_asterisk.c: (0x7f0c70023de0) RTP 0x7f0c70025530 -- Received packet from 178.62.121.41:19826, dropping due to strict RTP protection. [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:33] DEBUG[14593] bridge_channel.c: Bridge is returning 0x7f0c08021e90(SIP/zvonobot-00000002) to read format alaw [Aug 18 10:34:33] DEBUG[14593] channel.c: Channel SIP/zvonobot-00000002 setting read format path: alaw -> alaw [Aug 18 10:34:33] DEBUG[14593] bridge_channel.c: Bridge is returning 0x7f0c08021e90(SIP/zvonobot-00000002) to write format alaw [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213158': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[14593] channel.c: Channel SIP/zvonobot-00000002 setting write format path: alaw -> alaw [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213158' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[14593] stasis/control.c: 212966, 79f92216-f8f4-49dd-85f1-f154853e1fd1: Channel was departed from bridge [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:535/channel:213158, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:535/channel:213158': 0x7f0c3800a460 destroyed [Aug 18 10:34:33] DEBUG[14593] stasis/app.c: bridge '79f92216-f8f4-49dd-85f1-f154853e1fd1': is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:33] DEBUG[12874] stasis/control.c: 212966: Channel departing bridge [Aug 18 10:34:33] DEBUG[14593] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:33] DEBUG[12874] bridge.c: Waiting for 0x7f0c08021e90(SIP/zvonobot-00000002) bridge thread to die. [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:33] DEBUG[15347] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:33] DEBUG[12874] stasis/app.c: channel '212966': is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15116] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15116] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[12874] channel.c: Channel 0x7f0c100160e0 'SIP/zvonobot-00000002' hanging up. Refs: 3 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213138': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213138' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[13540] chan_sip.c: Hangup call SIP/zvonobot-00000046, SIP callid 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[13540] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13540] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13540] channel.c: Channel 0x7f0c9c03d0a0 'SIP/zvonobot-00000046' destroying [Aug 18 10:34:33] DEBUG[14888] channel.c: Channel 0x7f0c1003bdc0 'Recorder/ARI-00000040;2' destroying [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.684, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:450/channel:213138, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.684': 0x7f0c3003bd10 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:780/channel:1629282873.684, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:780/channel:1629282873.684': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[15248] stasis.c: Creating topic. name: channel:1629282873.685, detail: [Aug 18 10:34:33] DEBUG[15127] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15127] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:450/channel:213138': 0x7f0c2c0b34b0 destroyed [Aug 18 10:34:33] DEBUG[15248] stasis.c: Topic 'channel:1629282873.685': 0x7f0c38034ba0 created [Aug 18 10:34:33] DEBUG[15248] stasis.c: Creating topic. name: cache:781/channel:1629282873.685, detail: [Aug 18 10:34:33] DEBUG[15248] stasis.c: Topic 'cache:781/channel:1629282873.685': 0x7f0c3800a460 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:780/channel:1629282873.684, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213138, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:780/channel:1629282873.684': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213138': 0x7f0c2c0f2b40 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.684, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.684': 0x7f0c3003bd10 destroyed [Aug 18 10:34:33] DEBUG[15130] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15130] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[14888] stasis.c: Destroying topic. name: cache:532/channel:1629282860.462, detail: [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:19', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000b0', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213138', '')] [Aug 18 10:34:33] DEBUG[14888] stasis.c: Topic 'cache:532/channel:1629282860.462': 0x7f0c10065e40 destroyed [Aug 18 10:34:33] DEBUG[14888] stasis.c: Destroying topic. name: channel:1629282860.462, detail: [Aug 18 10:34:33] DEBUG[14888] stasis.c: Topic 'channel:1629282860.462': 0x7f0c100307d0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213158, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213158': 0x7f0c3803d650 destroyed [Aug 18 10:34:33] DEBUG[15347] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:33] DEBUG[15132] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15132] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.686, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.686': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:782/channel:1629282873.686, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:782/channel:1629282873.686': 0x7f0c3003bd10 created [Aug 18 10:34:33] DEBUG[15126] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:33] DEBUG[15126] http.c: HTTP closing session. Top level [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7;received=159.65.48.104 From: ;tag=as786713e2 To: ;tag=as51d52ab5 Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="408ab055" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[14011] res_rtp_asterisk.c: (0x7f0c240f8a30) DTLS stop [Aug 18 10:34:33] DEBUG[14011] res_rtp_asterisk.c: (0x7f0c240f8a30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[14011] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[14011] res_rtp_asterisk.c: (0x7f0c240f8a30) ICE stopped [Aug 18 10:34:33] DEBUG[14011] rtp_engine.c: Destroyed RTP instance '0x7f0c240f8a30' [Aug 18 10:34:33] DEBUG[14011] channel.c: Channel 0x7f0c240f6d50 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30' destroying [Aug 18 10:34:33] DEBUG[15133] channel.c: Channel 0x7f0c10106440 'Recorder/ARI-00000055;2' allocated [Aug 18 10:34:33] DEBUG[15133] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:33] DEBUG[15134] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15134] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[14132] channel.c: Channel 0x7f0c1c026d20 'SIP/zvonobot-00000013' destroying [Aug 18 10:34:33] DEBUG[15348] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: 0x7f0c1007a360(Recorder/ARI-00000055;2) is joining [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213011': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213011' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:67/channel:213011, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:67/channel:213011': 0x7f0c980241b0 destroyed [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[15348] bridge_channel.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: pushing 0x7f0c1007a360(Recorder/ARI-00000055;2) [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213011, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213011': 0x7f0c98025620 destroyed [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as786713e2 [Aug 18 10:34:33] DEBUG[14132] channel.c: Channel 0x7f0c88099000 'Snoop/212982-00000010' destroying [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as51d52ab5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="408ab055" [Aug 18 10:34:33] DEBUG[14132] stasis.c: Destroying topic. name: cache:293/channel:1629282845.251, detail: [Aug 18 10:34:33] DEBUG[14132] stasis.c: Topic 'cache:293/channel:1629282845.251': 0x7f0c880756b0 destroyed [Aug 18 10:34:33] DEBUG[14132] stasis.c: Destroying topic. name: channel:1629282845.251, detail: [Aug 18 10:34:33] DEBUG[14132] stasis.c: Topic 'channel:1629282845.251': 0x7f0c8800f890 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:782/channel:1629282873.686, detail: [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 454 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 454 [Aug 18 10:34:33] DEBUG[15348] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:782/channel:1629282873.686': 0x7f0c3003bd10 destroyed [Aug 18 10:34:33] VERBOSE[15348] bridge_channel.c: Channel Recorder/ARI-00000055;2 joined 'simple_bridge' stasis-bridge <5c24e2ba-8671-4745-b349-4500db0d3cb5> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 (Checking To) --From tag as786713e2 --To-tag as51d52ab5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116876@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2050cea7 Max-Forwards: 70 From: ;tag=as786713e2 To: Contact: Call-ID: 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[15348] bridge_native_rtp.c: Bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5'. Checking compatability for channels 'Recorder/ARI-00000041;2' and 'Recorder/ARI-00000055;2' [Aug 18 10:34:33] DEBUG[15348] bridge_native_rtp.c: Bridge '5c24e2ba-8671-4745-b349-4500db0d3cb5' can not use native RTP bridge as could not get details [Aug 18 10:34:33] DEBUG[15348] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15348] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (1) INVITE - 5 [Aug 18 10:34:33] DEBUG[15348] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06fea4c6 Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116856@178.62.121.41", nonce="4ecc103f", response="5984a684e9cf4d2ea38b7a835805bdb8" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029195 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:33] DEBUG[15348] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15348] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5 is already using the new technology. [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #59 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:33] DEBUG[15348] bridge.c: Bridge 5c24e2ba-8671-4745-b349-4500db0d3cb5: 0x7f0c1007a360(Recorder/ARI-00000055;2) is joining simple_bridge technology [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.686, detail: [Aug 18 10:34:33] DEBUG[20535] devicestate.c: Changing state for UnicastRTP/127.0.0.1:50194 - state 0 (Unknown) [Aug 18 10:34:33] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/UnicastRTP/127.0.0.1:50194, detail: [Aug 18 10:34:33] DEBUG[20535] stasis.c: Topic 'devicestate:all/UnicastRTP/127.0.0.1:50194': 0x7f0c84137f70 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #59)) [Aug 18 10:34:33] DEBUG[20616] app_queue.c: Device 'UnicastRTP/127.0.0.1:50194' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213037': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213037' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[15348] channel.c: Channel Recorder/ARI-00000055;2 setting read format path: slin -> slin [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116849@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK10cbf9d0 Max-Forwards: 70 From: ;tag=as1781187a To: Contact: Call-ID: 64e5898d41ad108f4bafedb86b292fc7@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 787513059 787513059 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18212 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.686': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for SIP - zvonobot [Aug 18 10:34:33] DEBUG[20535] chan_sip.c: Checking device state for peer zvonobot [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:171/channel:213037, detail: [Aug 18 10:34:33] DEBUG[20535] devicestate.c: Changing state for SIP/zvonobot - state 1 (Not in use) [Aug 18 10:34:33] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Snoop - 212986 [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:50', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000030', '', 'Stasis', 'calls_0', 38, 32, 'ANSWERED', 3, '', '213011', '')] [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:171/channel:213037': 0x7f0c9c044240 destroyed [Aug 18 10:34:33] DEBUG[20535] devicestate.c: Changing state for Snoop/212986 - state 4 (Invalid) [Aug 18 10:34:33] DEBUG[15348] channel.c: Channel Recorder/ARI-00000041;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #89 (1) INVITE - 5 [Aug 18 10:34:33] DEBUG[20535] stasis.c: Creating topic. name: devicestate:all/Snoop/212986, detail: [Aug 18 10:34:33] DEBUG[15151] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[20535] stasis.c: Topic 'devicestate:all/Snoop/212986': 0x7f0c84105960 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #89)) [Aug 18 10:34:33] DEBUG[15151] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213037, detail: [Aug 18 10:34:33] DEBUG[15348] channel.c: Channel Recorder/ARI-00000041;2 setting read format path: slin -> slin [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213037': 0x7f0c9c03dfb0 destroyed [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116830@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd Max-Forwards: 70 From: ;tag=as39b021ce To: Contact: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1100223183 1100223183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.687, detail: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.687': 0x7f0c3003bd10 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #39 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #39)) [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:783/channel:1629282873.687, detail: [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116832@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 Max-Forwards: 70 From: ;tag=as7025f96b To: Contact: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 58040179 58040179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16686 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:783/channel:1629282873.687': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[15349] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116828@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df Max-Forwards: 70 From: ;tag=as7b799277 To: Contact: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 670818967 670818967 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #25 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[15253] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[20616] app_queue.c: Device 'Snoop/212986' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:783/channel:1629282873.687, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:783/channel:1629282873.687': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.687, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.687': 0x7f0c3003bd10 destroyed [Aug 18 10:34:33] DEBUG[15253] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:03', '"" <>', '', 's', 'default', 'Snoop/213011-0000000f', 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30', 'Stasis', 'calls_0', 27, 27, 'ANSWERED', 3, '', '1629282842.223', '')] [Aug 18 10:34:33] DEBUG[20535] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Aug 18 10:34:33] DEBUG[15349] http.c: HTTP Request URI is /ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:34:33] DEBUG[15012] bridge_channel.c: Setting 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) state from:0 to:1 [Aug 18 10:34:33] DEBUG[15016] channel.c: Channel 0x7f0c8c106b00 'Recorder/ARI-00000046;1' destroying [Aug 18 10:34:33] DEBUG[15016] stasis.c: Destroying topic. name: cache:519/channel:1629282860.450, detail: [Aug 18 10:34:33] DEBUG[15016] stasis.c: Topic 'cache:519/channel:1629282860.450': 0x7f0c8c055680 destroyed [Aug 18 10:34:33] DEBUG[15016] stasis.c: Destroying topic. name: channel:1629282860.450, detail: [Aug 18 10:34:33] DEBUG[15016] stasis.c: Topic 'channel:1629282860.450': 0x7f0c8c048d20 destroyed [Aug 18 10:34:33] DEBUG[15350] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15350] http.c: HTTP Request URI is /ari/channels/1629282835.124 [Aug 18 10:34:33] DEBUG[15348] channel.c: Channel Recorder/ARI-00000055;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #25)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116833@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d Max-Forwards: 70 From: ;tag=as235fcd7f To: Contact: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1532262111 1532262111 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14362 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '1db0d1b33476bee66d27be26316d6a17@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:33] DEBUG[15350] http.c: match request [ari/channels/1629282835.124] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15350] http.c: match request [ari/channels/1629282835.124] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15349] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15012] bridge_channel.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: pulling 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) [Aug 18 10:34:33] VERBOSE[15012] bridge_channel.c: Channel Recorder/ARI-00000046;2 left 'simple_bridge' stasis-bridge <6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb> [Aug 18 10:34:33] DEBUG[15012] bridge_channel.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb: 0x7f0c8c0fd7e0(Recorder/ARI-00000046;2) is leaving simple_bridge technology [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:33] DEBUG[15350] http.c: match request [ari/channels/1629282835.124] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-00000002 - start 1629282822.069331 answer 1629282857.809594 end 1629282872.993870 dur 50.924 bill 15.184 dispo ANSWERED [Aug 18 10:34:33] DEBUG[15350] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15012] bridge_native_rtp.c: Bridge '6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb' can not use native RTP bridge as two channels are required [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.688, detail: [Aug 18 10:34:33] DEBUG[15159] stasis.c: Creating topic. name: channel:1629282873.689, detail: [Aug 18 10:34:33] DEBUG[15159] stasis.c: Topic 'channel:1629282873.689': 0x7f0c80058080 created [Aug 18 10:34:33] DEBUG[15159] stasis.c: Creating topic. name: cache:784/channel:1629282873.689, detail: [Aug 18 10:34:33] DEBUG[15159] stasis.c: Topic 'cache:784/channel:1629282873.689': 0x7f0c8003ccd0 created [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0c3c00) DTLS stop [Aug 18 10:34:33] DEBUG[15012] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15012] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15012] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.688': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_213011': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[15349] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[14825] stasis.c: Creating topic. name: channel:1629282873.690, detail: [Aug 18 10:34:33] DEBUG[14825] stasis.c: Topic 'channel:1629282873.690': 0x7f0c300b3540 created [Aug 18 10:34:33] DEBUG[14825] stasis.c: Creating topic. name: cache:786/channel:1629282873.690, detail: [Aug 18 10:34:33] DEBUG[14825] stasis.c: Topic 'cache:786/channel:1629282873.690': 0x7f0c3003bd10 created [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0c3c00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15147] channel.c: Channel 0x7f0c8408ee40 'Announcer/ARI-0000004a;2' destroying [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0c3c00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15012] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[14974] channel.c: Channel 0x2ca1be0 'Announcer/ARI-0000005c;1' allocated [Aug 18 10:34:33] DEBUG[14974] stasis.c: Creating topic. name: channel:1629282873.691, detail: [Aug 18 10:34:33] DEBUG[14974] stasis.c: Topic 'channel:1629282873.691': 0x2c3de20 created [Aug 18 10:34:33] DEBUG[15012] bridge.c: Bridge 6a98cdc9-a74a-4bb9-9be3-f2f08fa1b0fb is already using the new technology. [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Finding handler for channels/1629282835.124 [Aug 18 10:34:33] DEBUG[14974] stasis.c: Creating topic. name: cache:787/channel:1629282873.691, detail: [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_213011' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:276/channel:robot_213011, detail: [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:785/channel:1629282873.688, detail: [Aug 18 10:34:33] DEBUG[14974] stasis.c: Topic 'cache:787/channel:1629282873.691': 0x2c37290 created [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0c3c00) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c0c3c00' [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:276/channel:robot_213011': 0x7f0c24075da0 destroyed [Aug 18 10:34:33] DEBUG[15349] http.c: match request [ari/bridges/d36cece3-ab54-488a-bcb0-0ed40691a344] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15012] channel.c: Channel 0x7f0c8c03b720 'Recorder/ARI-00000046;2' hanging up. Refs: 2 [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213011, detail: [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:robot_213011': 0x7f0c24075fe0 destroyed [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Finding handler for 1629282835.124 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #4 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #4)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116865@178.62.121.41", nonce="328f7607", response="03e2ed289df65658183229073b55646c" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking channels create: Didn't match 1629282835.124 [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15350] res_ari.c: Checking channels externalMedia: Didn't match 1629282835.124 [Aug 18 10:34:33] DEBUG[15350] res_ari.c: No explicit handler found for 1629282835.124. Using wildcard channelId. [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '212982': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '212982' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:26/channel:212982, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:26/channel:212982': 0x7f0c1c028740 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:212982, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:212982': 0x7f0c1c028220 destroyed [Aug 18 10:34:33] DEBUG[15147] stasis.c: Destroying topic. name: cache:629/channel:1629282866.548, detail: [Aug 18 10:34:33] DEBUG[15147] stasis.c: Topic 'cache:629/channel:1629282866.548': 0x7f0c84049bf0 destroyed [Aug 18 10:34:33] DEBUG[15147] stasis.c: Destroying topic. name: channel:1629282866.548, detail: [Aug 18 10:34:33] DEBUG[15147] stasis.c: Topic 'channel:1629282866.548': 0x7f0c840780a0 destroyed [Aug 18 10:34:33] DEBUG[15349] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15006] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15006] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:785/channel:1629282873.688': 0x7f0c3002a5f0 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 30cb159f67289df002568fe9006f4752@159.65.48.104:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '30cb159f67289df002568fe9006f4752@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS stop [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Finding handler for bridges/d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:34:33] DEBUG[15165] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50497-0x7f0c98083570' [Aug 18 10:34:33] DEBUG[14756] res_rtp_asterisk.c: (0x7f0c38090cc0) DTLS stop [Aug 18 10:34:33] DEBUG[14756] res_rtp_asterisk.c: (0x7f0c38090cc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[14756] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[14756] res_rtp_asterisk.c: (0x7f0c38090cc0) ICE stopped [Aug 18 10:34:33] DEBUG[14756] rtp_engine.c: Destroyed RTP instance '0x7f0c38090cc0' [Aug 18 10:34:33] DEBUG[14756] channel.c: Channel 0x7f0c38071710 'UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0' destroying [Aug 18 10:34:33] DEBUG[15165] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:33] DEBUG[15165] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[14272] bridge_channel.c: Setting 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) state from:0 to:1 [Aug 18 10:34:33] DEBUG[14272] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: pulling 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) [Aug 18 10:34:33] VERBOSE[14272] bridge_channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 left 'simple_bridge' stasis-bridge [Aug 18 10:34:33] DEBUG[14272] bridge_channel.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe: 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) is leaving simple_bridge technology [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[14272] bridge_native_rtp.c: Bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe' can not use native RTP bridge as two channels are required [Aug 18 10:34:33] DEBUG[14272] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[14272] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[14272] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[14272] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[14272] bridge.c: Bridge a76fe935-dd52-4012-a523-638ab1ec4dfe is already using the new technology. [Aug 18 10:34:33] DEBUG[14272] bridge_channel.c: Bridge is returning 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) to write format slin16 [Aug 18 10:34:33] DEBUG[14272] channel.c: Channel UnicastRTP/127.0.0.1:50497-0x7f0c98083570 setting write format path: slin16 -> slin16 [Aug 18 10:34:33] DEBUG[14272] stasis/control.c: robot_212982, a76fe935-dd52-4012-a523-638ab1ec4dfe: Channel was departed from bridge [Aug 18 10:34:33] DEBUG[14272] stasis/app.c: bridge 'a76fe935-dd52-4012-a523-638ab1ec4dfe': is 2 interested in calls_0 [Aug 18 10:34:33] DEBUG[14215] stasis/control.c: robot_212982: Channel departing bridge [Aug 18 10:34:33] DEBUG[14215] bridge.c: Waiting for 0x7f0c180c3b90(UnicastRTP/127.0.0.1:50497-0x7f0c98083570) bridge thread to die. [Aug 18 10:34:33] DEBUG[15257] channel.c: Channel 0x7f0c84148b70 'Announcer/ARI-0000005d;1' allocated [Aug 18 10:34:33] DEBUG[15257] stasis.c: Creating topic. name: channel:1629282873.692, detail: [Aug 18 10:34:33] DEBUG[15257] stasis.c: Topic 'channel:1629282873.692': 0x7f0c8410d4b0 created [Aug 18 10:34:33] DEBUG[15257] stasis.c: Creating topic. name: cache:788/channel:1629282873.692, detail: [Aug 18 10:34:33] DEBUG[15257] stasis.c: Topic 'cache:788/channel:1629282873.692': 0x7f0c84125d90 created [Aug 18 10:34:33] DEBUG[14272] stasis/control.c: reason: Channel was departed from bridge [Aug 18 10:34:33] DEBUG[15351] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[14215] stasis/app.c: channel 'robot_212982': is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[14215] channel.c: Channel 0x7f0c9808fde0 'UnicastRTP/127.0.0.1:50497-0x7f0c98083570' hanging up. Refs: 2 [Aug 18 10:34:33] DEBUG[15351] http.c: HTTP Request URI is /ari/channels/212982 [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c2c0a9e10) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[15351] http.c: match request [ari/channels/212982] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15259] channel.c: Channel 0x7f0c8000f880 'UnicastRTP/127.0.0.1:50017-0x7f0c80021ed0' allocated [Aug 18 10:34:33] DEBUG[15133] res_stasis_recording.c: 1629282869.599: Sending record(213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd.wav) command [Aug 18 10:34:33] WARNING[15133] res_stasis_recording.c: Recording file '/var/spool/asterisk/recording/213021_WWxdRgfUAbUtkABDvUBnpaAdUOrYJXJd' already exists and ifExists option is failure. [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15259] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:33] DEBUG[15351] http.c: match request [ari/channels/212982] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15133] channel.c: Channel 0x7f0c10123540 'Recorder/ARI-00000055;1' hanging up. Refs: 2 [Aug 18 10:34:33] DEBUG[15133] autoservice.c: Thread is a user interface, not removing channel Recorder/ARI-00000055;1 from autoservice [Aug 18 10:34:33] DEBUG[15102] channel.c: Recorder/ARI-00000041;2: Dropping redundant connected line update "" <>. [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_212991': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_212991' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:429/channel:robot_212991, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:429/channel:robot_212991': 0x7f0c38066b90 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212991, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:robot_212991': 0x7f0c3807e8f0 destroyed [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:33] DEBUG[15348] channel.c: Recorder/ARI-00000055;2: Dropping redundant connected line update "" <>. [Aug 18 10:34:33] DEBUG[15351] http.c: match request [ari/channels/212982] with handler [ari] len 3 [Aug 18 10:34:33] VERBOSE[15259] res_rtp_asterisk.c: 0x7f0c8000cc80 -- Strict RTP learning after remote address set to: 127.0.0.1:50017 [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c2c0a9e10' [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15259] res_stasis.c: calls_0: Subscribing to robot_212984 [Aug 18 10:34:33] DEBUG[15259] stasis/app.c: Channel 'robot_212984' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15351] http.c: Match made with [ari] [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4090bbdc6ff80ba4244e03e744aa02c9@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:33] DEBUG[15259] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Finding handler for channels/212982 [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15259] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3808cb20) DTLS stop [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Finding handler for 212982 [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking channels create: Didn't match 212982 [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15351] res_ari.c: Checking channels externalMedia: Didn't match 212982 [Aug 18 10:34:33] DEBUG[15351] res_ari.c: No explicit handler found for 212982. Using wildcard channelId. [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3808cb20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3808cb20) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c3808cb20) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c3808cb20' [Aug 18 10:34:33] VERBOSE[15352] dial.c: Called 127.0.0.1:50017 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:785/channel:1629282873.688, detail: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 251d1fab55864c302358179000f7c9b8@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:785/channel:1629282873.688': 0x7f0c3002a5f0 destroyed [Aug 18 10:34:33] DEBUG[14394] res_rtp_asterisk.c: (0x7f0c38023850) RTCP got report of 100 bytes from 178.62.121.41:12659 [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '251d1fab55864c302358179000f7c9b8@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS stop [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c9c04bc00) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c9c04bc00' [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 From: ;tag=as7dd13c21 To: ;tag=as19c9362c Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK720164af;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7dd13c21 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as19c9362c [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2370dec7" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 (Checking To) --From tag as7dd13c21 --To-tag as19c9362c [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 61df87fc22a6dfff30a45a0426505a64@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b84ac14 Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116859@178.62.121.41", nonce="4cb8ffaa", response="34d7073f228c83199f1d8a15ca6aeca9" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116827@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d Max-Forwards: 70 From: ;tag=as568913af To: Contact: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 962154366 962154366 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #150 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #150)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116879@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6d8d001a Max-Forwards: 70 From: ;tag=as58ae887d To: Contact: Call-ID: 31daac6c3c7b4d211e20976a4d49ca0f@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116879@178.62.121.41", nonce="6c2e56a5", response="40a469e8ebc5c363b0147aac80172fab" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1671475453 1671475454 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16950 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26886e91 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116860@178.62.121.41", nonce="09aacdf1", response="3c5bdea99ab5d5c94251dcfc28700249" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890281 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06fea4c6 Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116856@178.62.121.41", nonce="4ecc103f", response="5984a684e9cf4d2ea38b7a835805bdb8" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029195 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #159 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #159)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116885@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4c11f6f4 Max-Forwards: 70 From: ;tag=as1cccf2a3 To: Contact: Call-ID: 0d1280065e8aab5644bcc84653498f56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116885@178.62.121.41", nonce="4a943c2f", response="1e6d5af191b81b509e50fc181f912f28" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1613790570 1613790571 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16374 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #28 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #28)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116864@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK437e89d6 Max-Forwards: 70 From: ;tag=as220fc8e1 To: Contact: Call-ID: 45bb066f2988fa763849c28f6566e41c@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116864@178.62.121.41", nonce="3f8b6b29", response="43c73b210726c419b446d559637382e6" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2034266556 2034266557 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16258 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116831@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 Max-Forwards: 70 From: ;tag=as655f7a70 To: Contact: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2053228906 2053228906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19488 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #127 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #127)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116836@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4b5dd0b6 Max-Forwards: 70 From: ;tag=as2d4671b8 To: Contact: Call-ID: 64d2aac75896f6d3140509d05f605000@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1289209748 1289209748 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.688, detail: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #89 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.688': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[12894] chan_sip.c: Hangup call SIP/zvonobot-00000006, SIP callid 32d9aa3a198e830d5e35f1553d21087d@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[12894] res_rtp_asterisk.c: (0x7f0c34009e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[12894] res_rtp_asterisk.c: (0x7f0c34009e10) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15193] channel.c: Channel 0x7f0c1802bc40 'Announcer/ARI-00000056;2' allocated [Aug 18 10:34:33] DEBUG[15193] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:33] DEBUG[15193] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000056;1' [Aug 18 10:34:33] DEBUG[15353] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c18090d60(Announcer/ARI-00000056;2) is joining [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15267] channel.c: Channel 0x7f0ca8076ba0 'Announcer/ARI-0000005e;1' allocated [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #89)) [Aug 18 10:34:33] DEBUG[15004] res_stasis.c: 1629282863.497: Hangup (no more frames) [Aug 18 10:34:33] DEBUG[15267] stasis.c: Creating topic. name: channel:1629282873.693, detail: [Aug 18 10:34:33] DEBUG[15267] stasis.c: Topic 'channel:1629282873.693': 0x7f0ca800e860 created [Aug 18 10:34:33] DEBUG[15267] stasis.c: Creating topic. name: cache:789/channel:1629282873.693, detail: [Aug 18 10:34:33] DEBUG[15267] stasis.c: Topic 'cache:789/channel:1629282873.693': 0x7f0ca80572b0 created [Aug 18 10:34:33] DEBUG[15007] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15353] bridge_channel.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: pushing 0x7f0c18090d60(Announcer/ARI-00000056;2) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116830@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd Max-Forwards: 70 From: ;tag=as39b021ce To: Contact: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1100223183 1100223183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[15004] stasis/app.c: channel '1629282863.497': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[15004] stasis/app.c: channel '1629282863.497' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[15007] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Finding handler for d36cece3-ab54-488a-bcb0-0ed40691a344 [Aug 18 10:34:33] DEBUG[15349] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:56', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000046', '', 'AppDial2', '(Outgoing Line)', 33, 0, 'BUSY', 3, '', '213037', '')] [Aug 18 10:34:33] DEBUG[15349] res_ari.c: No explicit handler found for d36cece3-ab54-488a-bcb0-0ed40691a344. Using wildcard bridgeId. [Aug 18 10:34:33] DEBUG[15004] channel.c: Channel 0x7f0ca40f4750 'Snoop/212969-00000017' hanging up. Refs: 3 [Aug 18 10:34:33] DEBUG[15353] bridge_roles.c: Set role 'announcer' [Aug 18 10:34:33] VERBOSE[15353] bridge_channel.c: Channel Announcer/ARI-00000056;2 joined 'simple_bridge' stasis-bridge <3fc9ee09-2746-49ab-833c-6c9b37b1bb83> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #75 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #75)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116878@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3dc016f5 Max-Forwards: 70 From: ;tag=as45eb6124 To: Contact: Call-ID: 3885c14f5099607a56cac7531a0f6e36@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116878@178.62.121.41", nonce="3e902bc1", response="1ddcb6d30cd03b1b81e6c99e0cc6adc5" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 585612348 585612349 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11726 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15349] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: telling all channels to leave the party [Aug 18 10:34:33] VERBOSE[15352] dial.c: UnicastRTP/127.0.0.1:50017-0x7f0c80021ed0 answered [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15349] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:33] DEBUG[15353] bridge.c: Chose bridge technology softmix [Aug 18 10:34:33] VERBOSE[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: switching from simple_bridge technology to softmix [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.694, detail: [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling softmix technology constructor [Aug 18 10:34:33] DEBUG[15269] channel.c: Channel 0x7f0c9c0a9be0 'UnicastRTP/127.0.0.1:50035-0x7f0c9c0696e0' allocated [Aug 18 10:34:33] DEBUG[15269] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:33] VERBOSE[15269] res_rtp_asterisk.c: 0x7f0c9c010020 -- Strict RTP learning after remote address set to: 127.0.0.1:50035 [Aug 18 10:34:33] DEBUG[15269] res_stasis.c: calls_0: Subscribing to robot_213038 [Aug 18 10:34:33] DEBUG[15269] stasis/app.c: Channel 'robot_213038' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15349] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: queueing action type:13 sub:1001 [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: moving 0x7f0c2c006860(SIP/zvonobot-00000033) to dummy bridge temporarily [Aug 18 10:34:33] DEBUG[15269] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.694': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:790/channel:1629282873.694, detail: [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling stasis bridge destructor [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology stop [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge d36cece3-ab54-488a-bcb0-0ed40691a344: calling simple_bridge technology destructor [Aug 18 10:34:33] DEBUG[15349] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:33] DEBUG[15349] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: moving 0x7f0c780a94c0(Recorder/ARI-0000002d;2) to dummy bridge temporarily [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c2c006860(SIP/zvonobot-00000033) is leaving simple_bridge technology (dummy) [Aug 18 10:34:33] DEBUG[15356] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15287] channel.c: Channel 0x7f0c200b0230 'UnicastRTP/127.0.0.1:50267-0x7f0c2001ddd0' allocated [Aug 18 10:34:33] DEBUG[15287] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:33] VERBOSE[15287] res_rtp_asterisk.c: 0x7f0c200171e0 -- Strict RTP learning after remote address set to: 127.0.0.1:50267 [Aug 18 10:34:33] DEBUG[13993] res_rtp_asterisk.c: (0x7f0ca003db80) DTLS stop [Aug 18 10:34:33] DEBUG[13993] res_rtp_asterisk.c: (0x7f0ca003db80) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13993] res_rtp_asterisk.c: (0x7f0ca003db80) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[13341] res_rtp_asterisk.c: (0x7f0ca406c1b0) DTLS stop [Aug 18 10:34:33] DEBUG[13341] res_rtp_asterisk.c: (0x7f0ca406c1b0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13341] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[13341] res_rtp_asterisk.c: (0x7f0ca406c1b0) ICE stopped [Aug 18 10:34:33] DEBUG[13341] rtp_engine.c: Destroyed RTP instance '0x7f0ca406c1b0' [Aug 18 10:34:33] DEBUG[13341] channel.c: Channel 0x7f0ca40752f0 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0' destroying [Aug 18 10:34:33] DEBUG[13993] res_rtp_asterisk.c: (0x7f0ca003db80) ICE stopped [Aug 18 10:34:33] DEBUG[13993] rtp_engine.c: Destroyed RTP instance '0x7f0ca003db80' [Aug 18 10:34:33] DEBUG[13993] channel.c: Channel 0x7f0ca00dd400 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80' destroying [Aug 18 10:34:33] DEBUG[15269] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:790/channel:1629282873.694': 0x7f0c300fd3c0 created [Aug 18 10:34:33] DEBUG[15213] channel.c: Channel 0x7f0c88037560 'Recorder/ARI-00000059;2' allocated [Aug 18 10:34:33] DEBUG[15059] channel.c: Channel 0x7f0c7404bf40 'Recorder/ARI-00000053;2' allocated [Aug 18 10:34:33] DEBUG[14912] chan_sip.c: Hangup call SIP/zvonobot-000000c8, SIP callid 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[14912] res_rtp_asterisk.c: (0x7f0c10088ef0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[14912] res_rtp_asterisk.c: (0x7f0c10088ef0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[14912] channel.c: Channel 0x7f0c100480d0 'SIP/zvonobot-000000c8' destroying [Aug 18 10:34:33] DEBUG[15356] http.c: HTTP Request URI is /ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:34:33] DEBUG[15059] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[15208] channel.c: Channel 0x7f0c780a4330 'Recorder/ARI-00000057;2' allocated [Aug 18 10:34:33] DEBUG[15208] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[13785] channel.c: Channel 0x7f0c7c03c750 'SIP/zvonobot-0000003b' destroying [Aug 18 10:34:33] DEBUG[13785] channel.c: Channel 0x7f0c180b81a0 'Snoop/213023-0000000c' destroying [Aug 18 10:34:33] DEBUG[15198] channel.c: Channel 0x7f0c2c0bc390 'Recorder/ARI-00000058;2' allocated [Aug 18 10:34:33] DEBUG[15198] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[15356] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15284] channel.c: Channel 0x7f0c24143960 'Announcer/ARI-0000005f;1' allocated [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c780a94c0(Recorder/ARI-0000002d;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:33] VERBOSE[15352] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50017-0x7f0c80021ed0 [Aug 18 10:34:33] DEBUG[15021] channel.c: Channel 0x7f0ca411dd00 'Snoop/213041-0000001e' allocated [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:790/channel:1629282873.694, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:790/channel:1629282873.694': 0x7f0c300fd3c0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.694, detail: [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling simple_bridge technology stop [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c18090d60(Announcer/ARI-00000056;2) is joining softmix technology [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Announcer/ARI-00000056;2: [Aug 18 10:34:33] DEBUG[15353] channel.c: Channel Announcer/ARI-00000056;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Announcer/ARI-00000056;2: [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Announcer/ARI-00000056;2: Not in SFU mode [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c2c006860(SIP/zvonobot-00000033) is joining softmix technology [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: SIP/zvonobot-00000033: [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15357] bridge_channel.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: 0x7f0c78074930(Recorder/ARI-00000057;2) is joining [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051;received=159.65.48.104 From: ;tag=as3e829f44 To: ;tag=as4c9212d3 Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e1715ed" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51231051;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3e829f44 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as4c9212d3 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e1715ed" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 (Checking To) --From tag as3e829f44 --To-tag as4c9212d3 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 32cbddd62673253d5fac257e298c2963@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #29 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #29)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116883@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK4e310fc0 Max-Forwards: 70 From: ;tag=as69c6d00c To: Contact: Call-ID: 58e0aa954712c9223aa2bb5f2dc4df26@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116883@178.62.121.41", nonce="67379e21", response="bf564aaeaba47354a704299b969ea89a" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1127864104 1127864105 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16210 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.694': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:05', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50349-0x7f0c240f8a30', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213011', '')] [Aug 18 10:34:33] DEBUG[15356] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15356] http.c: match request [ari/bridges/4918ac35-38b0-4486-b626-7cf67dacf45b] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15356] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Finding handler for bridges/4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 446 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 446 [Aug 18 10:34:33] DEBUG[15021] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15284] stasis.c: Creating topic. name: channel:1629282873.695, detail: [Aug 18 10:34:33] DEBUG[15284] stasis.c: Topic 'channel:1629282873.695': 0x7f0c2403f610 created [Aug 18 10:34:33] DEBUG[15284] stasis.c: Creating topic. name: cache:791/channel:1629282873.695, detail: [Aug 18 10:34:33] DEBUG[15284] stasis.c: Topic 'cache:791/channel:1629282873.695': 0x7f0c24075fe0 created [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: SIP/zvonobot-00000033: [Aug 18 10:34:33] DEBUG[14685] bridge_native_rtp.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03'. Checking compatability for channels 'SIP/zvonobot-0000004c' and 'Recorder/ARI-00000045;2' [Aug 18 10:34:33] DEBUG[14685] bridge_native_rtp.c: Bridge '804476f3-9df1-4495-8c76-b406f7162d03' can not use native RTP bridge as channel 'SIP/zvonobot-0000004c' has features which prevent it [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: SIP/zvonobot-00000033: Not in SFU mode [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Finding handler for 4918ac35-38b0-4486-b626-7cf67dacf45b [Aug 18 10:34:33] DEBUG[15356] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15356] res_ari.c: No explicit handler found for 4918ac35-38b0-4486-b626-7cf67dacf45b. Using wildcard bridgeId. [Aug 18 10:34:33] DEBUG[15356] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: telling all channels to leave the party [Aug 18 10:34:33] DEBUG[15356] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: dissolving bridge with cause 16(Normal Clearing) [Aug 18 10:34:33] DEBUG[15356] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: queueing action type:13 sub:1001 [Aug 18 10:34:33] DEBUG[15356] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:33] DEBUG[15361] bridge_channel.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: 0x7f0c74049d20(Recorder/ARI-00000053;2) is joining [Aug 18 10:34:33] DEBUG[15361] bridge_channel.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: pushing 0x7f0c74049d20(Recorder/ARI-00000053;2) [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: actually destroying stasis bridge, nobody wants it anymore [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: calling stasis bridge destructor [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: calling simple_bridge technology stop [Aug 18 10:34:33] DEBUG[20534] bridge.c: Bridge 4918ac35-38b0-4486-b626-7cf67dacf45b: calling simple_bridge technology destructor [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: 0x7f0c780a94c0(Recorder/ARI-0000002d;2) is joining softmix technology [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Recorder/ARI-0000002d;2: [Aug 18 10:34:33] DEBUG[15353] channel.c: Channel Recorder/ARI-0000002d;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15353] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Recorder/ARI-0000002d;2: [Aug 18 10:34:33] DEBUG[15353] bridge_softmix.c: Recorder/ARI-0000002d;2: Not in SFU mode [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling softmix technology start [Aug 18 10:34:33] DEBUG[15353] bridge.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: calling simple_bridge technology destructor [Aug 18 10:34:33] DEBUG[14685] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] VERBOSE[15354] dial.c: Called 127.0.0.1:50035 [Aug 18 10:34:33] DEBUG[15213] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[15021] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15287] res_stasis.c: calls_0: Subscribing to robot_213027 [Aug 18 10:34:33] DEBUG[14685] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[13785] stasis.c: Destroying topic. name: cache:237/channel:1629282840.199, detail: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[14685] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15356] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.696, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.696': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:792/channel:1629282873.696, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:792/channel:1629282873.696': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[15358] bridge_channel.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: 0x7f0c2c0e3920(Recorder/ARI-00000058;2) is joining [Aug 18 10:34:33] DEBUG[14685] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '2cbf2a7b3b2272d37ec48fc03784df5a@159.65.48.104:5060' Method: INVITE [Aug 18 10:34:33] DEBUG[15287] stasis/app.c: Channel 'robot_213027' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[14685] bridge.c: Bridge 804476f3-9df1-4495-8c76-b406f7162d03 is already using the new technology. [Aug 18 10:34:33] DEBUG[13785] stasis.c: Topic 'cache:237/channel:1629282840.199': 0x7f0c180b95f0 destroyed [Aug 18 10:34:33] VERBOSE[15369] dial.c: Called 127.0.0.1:50267 [Aug 18 10:34:33] VERBOSE[15369] dial.c: UnicastRTP/127.0.0.1:50267-0x7f0c2001ddd0 answered [Aug 18 10:34:33] VERBOSE[15369] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50267-0x7f0c2001ddd0 [Aug 18 10:34:33] DEBUG[15369] stasis/app.c: Channel 'robot_213027' is 2 interested in calls_0 [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10088ef0) DTLS stop [Aug 18 10:34:33] DEBUG[15365] http.c: HTTP opening session. Top level [Aug 18 10:34:33] VERBOSE[15354] dial.c: UnicastRTP/127.0.0.1:50035-0x7f0c9c0696e0 answered [Aug 18 10:34:33] VERBOSE[15354] ari/resource_channels.c: Launching Stasis(calls_0) on UnicastRTP/127.0.0.1:50035-0x7f0c9c0696e0 [Aug 18 10:34:33] DEBUG[15354] stasis/app.c: Channel 'robot_213038' is 2 interested in calls_0 [Aug 18 10:34:33] DEBUG[14685] audiohook.c: Audiohook 0x7f0ca41196b0 has stale audio in its factories. Flushing them both [Aug 18 10:34:33] DEBUG[13785] stasis.c: Destroying topic. name: channel:1629282840.199, detail: [Aug 18 10:34:33] DEBUG[13785] stasis.c: Topic 'channel:1629282840.199': 0x7f0c18091270 destroyed [Aug 18 10:34:33] DEBUG[15361] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] VERBOSE[15361] bridge_channel.c: Channel Recorder/ARI-00000053;2 joined 'simple_bridge' stasis-bridge <466c6117-1cbd-4c34-863b-5d0db95ca0e0> [Aug 18 10:34:33] DEBUG[15287] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15365] http.c: HTTP Request URI is /ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/play?media=sound%3Asilence%2F2 [Aug 18 10:34:33] DEBUG[15368] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c880d2760(Recorder/ARI-00000059;2) is joining [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10088ef0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15365] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/play] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15287] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15365] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/play] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15365] http.c: match request [ari/bridges/804476f3-9df1-4495-8c76-b406f7162d03/play] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15365] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:792/channel:1629282873.696, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:792/channel:1629282873.696': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.696, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.696': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:44', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000013', '', 'Stasis', 'calls_0', 45, 29, 'ANSWERED', 3, '', '212982', '')] [Aug 18 10:34:33] DEBUG[15367] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Finding handler for bridges/804476f3-9df1-4495-8c76-b406f7162d03/play [Aug 18 10:34:33] DEBUG[15367] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213041&app=calls_0&format=slin16&external_host=127.0.0.1%3A50132 [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10088ef0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15355] bridge_softmix.c: Bridge 3fc9ee09-2746-49ab-833c-6c9b37b1bb83: starting mixing thread [Aug 18 10:34:33] DEBUG[15193] res_stasis_playback.c: 1629282870.605: Sending play(sound:silence/2) command [Aug 18 10:34:33] DEBUG[15357] bridge_channel.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: pushing 0x7f0c78074930(Recorder/ARI-00000057;2) [Aug 18 10:34:33] DEBUG[15357] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] VERBOSE[15357] bridge_channel.c: Channel Recorder/ARI-00000057;2 joined 'simple_bridge' stasis-bridge <8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95> [Aug 18 10:34:33] DEBUG[15357] bridge_native_rtp.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95'. Checking compatability for channels 'SIP/zvonobot-0000004d' and 'Recorder/ARI-00000057;2' [Aug 18 10:34:33] DEBUG[15357] bridge_native_rtp.c: Bridge '8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95' can not use native RTP bridge as could not get details [Aug 18 10:34:33] DEBUG[15357] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15357] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15357] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15357] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15357] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95 is already using the new technology. [Aug 18 10:34:33] DEBUG[15357] bridge.c: Bridge 8d3edf7f-fa9e-41a2-9ce5-f4d7e02cee95: 0x7f0c78074930(Recorder/ARI-00000057;2) is joining simple_bridge technology [Aug 18 10:34:33] DEBUG[15357] channel.c: Channel Recorder/ARI-00000057;2 setting read format path: slin -> slin [Aug 18 10:34:33] DEBUG[15357] channel.c: Channel SIP/zvonobot-0000004d setting write format path: slin -> alaw [Aug 18 10:34:33] DEBUG[15357] channel.c: Channel SIP/zvonobot-0000004d setting read format path: alaw -> slin [Aug 18 10:34:33] DEBUG[15357] channel.c: Channel Recorder/ARI-00000057;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c10088ef0) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.697, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.697': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:793/channel:1629282873.697, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:793/channel:1629282873.697': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:793/channel:1629282873.697, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:793/channel:1629282873.697': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.697, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.697': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:07', '"" <>', '', 's', 'default', 'Snoop/212982-00000010', 'UnicastRTP/127.0.0.1:50497-0x7f0c98083570', 'Stasis', 'calls_0', 23, 23, 'ANSWERED', 3, '', '1629282845.251', '')] [Aug 18 10:34:33] DEBUG[15193] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15208] res_stasis_recording.c: 1629282870.611: Sending record(213043_egRKVPBnNeSLLBzTdbfjfjBSRIkzUZEb.wav) command [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15368] bridge_channel.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: pushing 0x7f0c880d2760(Recorder/ARI-00000059;2) [Aug 18 10:34:33] DEBUG[15193] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15208] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15208] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: bridge 'd36cece3-ab54-488a-bcb0-0ed40691a344' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:107/bridge:d36cece3-ab54-488a-bcb0-0ed40691a344, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:107/bridge:d36cece3-ab54-488a-bcb0-0ed40691a344': 0x7f0c7004a970 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: bridge:d36cece3-ab54-488a-bcb0-0ed40691a344, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'bridge:d36cece3-ab54-488a-bcb0-0ed40691a344': 0x7f0c70030fe0 destroyed [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_212981': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_212981' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:115/channel:robot_212981, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:115/channel:robot_212981': 0x7f0ca4073500 destroyed [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 522 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 522 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_213009': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel 'robot_213009' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:273/channel:robot_213009, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:273/channel:robot_213009': 0x7f0ca00dfda0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_213009, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:robot_213009': 0x7f0ca00df2d0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:robot_212981, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:robot_212981': 0x7f0ca40766b0 destroyed [Aug 18 10:34:33] DEBUG[15367] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15358] bridge_channel.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: pushing 0x7f0c2c0e3920(Recorder/ARI-00000058;2) [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15372] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[14112] audiohook.c: Audiohook 0x7f0c8c051450 has stale audio in its factories. Flushing them both [Aug 18 10:34:33] DEBUG[15367] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Finding handler for 804476f3-9df1-4495-8c76-b406f7162d03 [Aug 18 10:34:33] DEBUG[14112] audiohook.c: Audiohook 0x7f0c1014c260 has stale audio in its factories. Flushing them both [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15367] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c10088ef0' [Aug 18 10:34:33] DEBUG[15367] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15367] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:33] DEBUG[15367] netsock2.c: Splitting '127.0.0.1:50132' into... [Aug 18 10:34:33] DEBUG[15367] netsock2.c: ...host '127.0.0.1' and port '50132'. [Aug 18 10:34:33] DEBUG[15367] netsock2.c: Splitting '127.0.0.1:50132' into... [Aug 18 10:34:33] DEBUG[15367] netsock2.c: ...host '127.0.0.1' and port '50132'. [Aug 18 10:34:33] DEBUG[15367] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:33] DEBUG[15367] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca4115480' [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) RTP allocated port 10028 [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) ICE creating session 127.0.0.1:10028 (10028) [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) ICE create [Aug 18 10:34:33] DEBUG[14112] res_rtp_asterisk.c: (0x7f0c24032c40) RTP ooh, format changed from none to ulaw [Aug 18 10:34:33] DEBUG[15365] res_ari.c: No explicit handler found for 804476f3-9df1-4495-8c76-b406f7162d03. Using wildcard bridgeId. [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Finding handler for play [Aug 18 10:34:33] DEBUG[20545] cdr.c: Finalized CDR for UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0 - start 1629282861.626422 answer 1629282861.763977 end 1629282873.203769 dur 11.577 bill 11.439 dispo ANSWERED [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.698, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.698': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:794/channel:1629282873.698, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:794/channel:1629282873.698': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:794/channel:1629282873.698, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:794/channel:1629282873.698': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.698, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.698': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:21', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50211-0x7f0c38090cc0', '', 'Stasis', 'calls_0', 11, 11, 'ANSWERED', 3, '', 'robot_212991', '')] [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15361] bridge_native_rtp.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0'. Checking compatability for channels 'SIP/zvonobot-00000040' and 'Recorder/ARI-00000053;2' [Aug 18 10:34:33] DEBUG[15361] bridge_native_rtp.c: Bridge '466c6117-1cbd-4c34-863b-5d0db95ca0e0' can not use native RTP bridge as could not get details [Aug 18 10:34:33] DEBUG[15371] app.c: play_and_record: , /var/spool/asterisk/recording/213043_egRKVPBnNeSLLBzTdbfjfjBSRIkzUZEb, 'wav' [Aug 18 10:34:33] DEBUG[15372] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) ICE add system candidates [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:33] DEBUG[15361] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:33] DEBUG[15365] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:33] DEBUG[15365] stasis.c: Creating topic. name: channel:1629282873.699, detail: [Aug 18 10:34:33] DEBUG[15361] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15371] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:33] DEBUG[15365] stasis.c: Topic 'channel:1629282873.699': 0x7f0ca010c820 created [Aug 18 10:34:33] DEBUG[15365] stasis.c: Creating topic. name: cache:795/channel:1629282873.699, detail: [Aug 18 10:34:33] DEBUG[15365] stasis.c: Topic 'cache:795/channel:1629282873.699': 0x7f0ca00eaed0 created [Aug 18 10:34:33] DEBUG[15352] stasis/app.c: Channel 'robot_212984' is 2 interested in calls_0 [Aug 18 10:34:33] VERBOSE[15371] app.c: x=0, open writing: /var/spool/asterisk/recording/213043_egRKVPBnNeSLLBzTdbfjfjBSRIkzUZEb format: wav, 0x2c55b20 [Aug 18 10:34:33] DEBUG[15368] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] VERBOSE[15368] bridge_channel.c: Channel Recorder/ARI-00000059;2 joined 'simple_bridge' stasis-bridge <724f7ab9-ed85-4748-9bb7-91218a7c6261> [Aug 18 10:34:33] DEBUG[15252] app.c: One waitfor failed, trying another [Aug 18 10:34:33] DEBUG[15361] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15361] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15361] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0 is already using the new technology. [Aug 18 10:34:33] DEBUG[15372] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213164': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[15361] bridge.c: Bridge 466c6117-1cbd-4c34-863b-5d0db95ca0e0: 0x7f0c74049d20(Recorder/ARI-00000053;2) is joining simple_bridge technology [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213164' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[15368] bridge.c: Chose bridge technology softmix [Aug 18 10:34:33] DEBUG[15358] bridge_roles.c: Set role 'recorder' [Aug 18 10:34:33] DEBUG[15367] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:33] DEBUG[15367] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) ICE add candidate: 159.65.48.104:10028, 2130706431 [Aug 18 10:34:33] DEBUG[15367] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:33] DEBUG[15367] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:33] DEBUG[15367] res_rtp_asterisk.c: (0x7f0ca4115480) ICE add candidate: 10.131.0.10:10028, 2130706431 [Aug 18 10:34:33] DEBUG[15367] rtp_engine.c: RTP instance '0x7f0ca4115480' is setup and ready to go [Aug 18 10:34:33] DEBUG[15367] stasis.c: Creating topic. name: channel:robot_213041, detail: [Aug 18 10:34:33] DEBUG[15367] stasis.c: Topic 'channel:robot_213041': 0x7f0ca406a890 created [Aug 18 10:34:33] DEBUG[15367] stasis.c: Creating topic. name: cache:796/channel:robot_213041, detail: [Aug 18 10:34:33] DEBUG[15367] stasis.c: Topic 'cache:796/channel:robot_213041': 0x7f0ca4100680 created [Aug 18 10:34:33] VERBOSE[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: switching from simple_bridge technology to softmix [Aug 18 10:34:33] DEBUG[15370] channel.c: Channel Announcer/ARI-00000056;1 setting write format path: gsm -> slin [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling softmix technology constructor [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel Recorder/ARI-00000053;2 setting read format path: slin -> slin [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: moving 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) to dummy bridge temporarily [Aug 18 10:34:33] DEBUG[15372] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: moving 0x23a5040(Recorder/ARI-00000044;2) to dummy bridge temporarily [Aug 18 10:34:33] DEBUG[15298] channel.c: Channel 0x7f0c7c073950 'SIP/zvonobot-000000fa' allocated [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:544/channel:213164, detail: [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) is leaving simple_bridge technology (dummy) [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x23a5040(Recorder/ARI-00000044;2) is leaving simple_bridge technology (dummy) [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling simple_bridge technology stop [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c880d2760(Recorder/ARI-00000059;2) is joining softmix technology [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000059;2: [Aug 18 10:34:33] DEBUG[15368] channel.c: Channel Recorder/ARI-00000059;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000059;2: [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000059;2: Not in SFU mode [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x7f0c7c0c86c0(SIP/zvonobot-0000003d) is joining softmix technology [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: SIP/zvonobot-0000003d: [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: SIP/zvonobot-0000003d: [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: SIP/zvonobot-0000003d: Not in SFU mode [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: 0x23a5040(Recorder/ARI-00000044;2) is joining softmix technology [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000044;2: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:544/channel:213164': 0x7f0c100737e0 destroyed [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:33] DEBUG[15298] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:33] DEBUG[15368] channel.c: Channel Recorder/ARI-00000044;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.701, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.701': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:797/channel:1629282873.701, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:797/channel:1629282873.701': 0x7f0c300413e0 created [Aug 18 10:34:33] VERBOSE[15358] bridge_channel.c: Channel Recorder/ARI-00000058;2 joined 'simple_bridge' stasis-bridge [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Aug 18 10:34:33] DEBUG[15368] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 512 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 512 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213023': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213023' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:96/channel:213023, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:96/channel:213023': 0x7f0c7c03f5a0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213023, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213023': 0x7f0c7c03df90 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: bridge '4918ac35-38b0-4486-b626-7cf67dacf45b': is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: bridge '4918ac35-38b0-4486-b626-7cf67dacf45b' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:121/bridge:4918ac35-38b0-4486-b626-7cf67dacf45b, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:121/bridge:4918ac35-38b0-4486-b626-7cf67dacf45b': 0x7f0c3c009390 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: bridge:4918ac35-38b0-4486-b626-7cf67dacf45b, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'bridge:4918ac35-38b0-4486-b626-7cf67dacf45b': 0x7f0c3c0305a0 destroyed [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000044;2: [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel SIP/zvonobot-00000040 setting write format path: slin -> ulaw [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel SIP/zvonobot-00000040 setting read format path: ulaw -> slin [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel Recorder/ARI-00000053;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15372] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15298] res_stasis.c: calls_0: Subscribing to 213215 [Aug 18 10:34:33] DEBUG[15298] stasis/app.c: Channel '213215' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15370] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 18 10:34:33] DEBUG[15368] bridge_softmix.c: Recorder/ARI-00000044;2: Not in SFU mode [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Outgoing Call for 79821116825 [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling softmix technology start [Aug 18 10:34:33] DEBUG[15368] bridge.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: calling simple_bridge technology destructor [Aug 18 10:34:33] VERBOSE[15370] file.c: Playing 'silence/2.gsm' (language 'en') [Aug 18 10:34:33] DEBUG[15359] stasis/app.c: Channel '1629282871.657' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15298] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15298] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15372] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:33] DEBUG[15359] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 82 instead [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213164, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213164': 0x7f0c10072d60 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:797/channel:1629282873.701, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:797/channel:1629282873.701': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.701, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.701': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:53', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50432-0x7f0ca406c1b0', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_212981', '')] [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.702, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.702': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:798/channel:1629282873.702, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:798/channel:1629282873.702': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:798/channel:1629282873.702, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:798/channel:1629282873.702': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.702, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.702': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:04', '"" <>', '', 's', 'default', 'UnicastRTP/127.0.0.1:50264-0x7f0ca003db80', '', 'Stasis', 'calls_0', 0, 0, 'ANSWERED', 3, '', 'robot_213009', '')] [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 528 [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 528 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:33] DEBUG[15059] res_stasis_recording.c: 1629282867.574: Sending record(213029_wGlDyvoiSvQdgQwKfOVYDJUhnGMUAKBe.wav) command [Aug 18 10:34:33] DEBUG[15059] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:33] DEBUG[15059] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] DEBUG[15358] bridge_native_rtp.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34'. Checking compatability for channels 'SIP/zvonobot-0000003f' and 'Recorder/ARI-00000058;2' [Aug 18 10:34:33] DEBUG[15316] channel.c: Channel 0x7f0c9c009830 'SIP/zvonobot-000000fb' allocated [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:33] DEBUG[15316] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:33] DEBUG[15376] http.c: HTTP opening session. Top level [Aug 18 10:34:33] VERBOSE[15374] chan_sip.c: Audio is at 17982 [Aug 18 10:34:33] DEBUG[15359] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15213] res_stasis_recording.c: 1629282870.618: Sending record(213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr.wav) command [Aug 18 10:34:33] WARNING[15213] res_stasis_recording.c: Recording file '/var/spool/asterisk/recording/213024_MqaWZizNxTtXzosCLvOnvtBxYWZGoFdr' already exists and ifExists option is failure. [Aug 18 10:34:33] DEBUG[15213] channel.c: Channel 0x7f0c8806f020 'Recorder/ARI-00000059;1' hanging up. Refs: 2 [Aug 18 10:34:33] DEBUG[15213] autoservice.c: Thread is a user interface, not removing channel Recorder/ARI-00000059;1 from autoservice [Aug 18 10:34:33] VERBOSE[15374] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] VERBOSE[15374] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[15374] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[15376] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:33] DEBUG[15373] bridge_softmix.c: Bridge 724f7ab9-ed85-4748-9bb7-91218a7c6261: starting mixing thread [Aug 18 10:34:33] DEBUG[15316] res_stasis.c: calls_0: Subscribing to 213219 [Aug 18 10:34:33] DEBUG[15316] stasis/app.c: Channel '213219' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15358] bridge_native_rtp.c: Bridge 'ef2f0ed6-7b56-46c3-a894-dc0114b2fb34' can not use native RTP bridge as could not get details [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 532 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 532 [Aug 18 10:34:33] DEBUG[15376] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15316] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15316] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15375] app.c: play_and_record: , /var/spool/asterisk/recording/213029_wGlDyvoiSvQdgQwKfOVYDJUhnGMUAKBe, 'wav' [Aug 18 10:34:33] DEBUG[15375] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:33] VERBOSE[15375] app.c: x=0, open writing: /var/spool/asterisk/recording/213029_wGlDyvoiSvQdgQwKfOVYDJUhnGMUAKBe format: wav, 0x7f0c180b7230 [Aug 18 10:34:33] DEBUG[15358] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15376] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15376] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15358] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Outgoing Call for 79821116821 [Aug 18 10:34:33] DEBUG[15358] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15358] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15358] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34 is already using the new technology. [Aug 18 10:34:33] DEBUG[15358] bridge.c: Bridge ef2f0ed6-7b56-46c3-a894-dc0114b2fb34: 0x7f0c2c0e3920(Recorder/ARI-00000058;2) is joining simple_bridge technology [Aug 18 10:34:33] DEBUG[15358] channel.c: Channel Recorder/ARI-00000058;2 setting read format path: slin -> slin [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 From: ;tag=as6be1179a To: ;tag=as34a9f263 Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 CSeq: 104 BYE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call leg/transaction does not exist [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Initializing initreq for method INVITE - callid 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116825@178.62.121.41 SIP/2.0 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK608439d6 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 3 [ 52]: From: ;tag=as5d5338d2 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 6 [ 60]: Call-ID: 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:33 GMT [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] VERBOSE[15374] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116825@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK608439d6 Max-Forwards: 70 From: ;tag=as5d5338d2 To: Contact: Call-ID: 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 479054008 479054008 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17982 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #118 [Aug 18 10:34:33] DEBUG[15374] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK01da8f12;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[15376] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as6be1179a [Aug 18 10:34:33] DEBUG[15358] channel.c: Channel SIP/zvonobot-0000003f setting write format path: slin -> alaw [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:33] DEBUG[14641] res_rtp_asterisk.c: (0x7f0c18094150) RTP ooh, format changed from none to ulaw [Aug 18 10:34:33] VERBOSE[15374] dial.c: Called zvonobot/79821116825 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as34a9f263 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.703, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.703': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:799/channel:1629282873.703, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:799/channel:1629282873.703': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:799/channel:1629282873.703, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:799/channel:1629282873.703': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.703, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.703': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:25', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-000000c8', '', 'AppDial2', '(Outgoing Line)', 6, 0, 'NO ANSWER', 3, '', '213164', '')] [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15358] channel.c: Channel SIP/zvonobot-0000003f setting read format path: alaw -> slin [Aug 18 10:34:33] DEBUG[15358] channel.c: Channel Recorder/ARI-00000058;2 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel Recorder/ARI-00000053;2 changing write format from slin to alaw, native formats (slin) [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15372] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15372] stasis.c: Creating topic. name: bridge:76c72761-45c7-42a4-9506-6bc68c2e0a5f, detail: [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:33] DEBUG[15378] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15372] stasis.c: Topic 'bridge:76c72761-45c7-42a4-9506-6bc68c2e0a5f': 0x7f0c101233b0 created [Aug 18 10:34:33] DEBUG[15372] stasis.c: Creating topic. name: cache:800/bridge:76c72761-45c7-42a4-9506-6bc68c2e0a5f, detail: [Aug 18 10:34:33] DEBUG[15372] stasis.c: Topic 'cache:800/bridge:76c72761-45c7-42a4-9506-6bc68c2e0a5f': 0x7f0c10036b90 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[15378] http.c: HTTP Request URI is /ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play?media=sound%3A%2Fvar%2Fwww%2Fstorage%2Faudio%2Fd4u0rPs9XhnEDOcN42vJ6M34KarentDf3ocOMkZc12XPZLEe6sPrO0eficvV7cq4 [Aug 18 10:34:33] DEBUG[15372] bridge_native_rtp.c: Bridge '76c72761-45c7-42a4-9506-6bc68c2e0a5f' can not use native RTP bridge as two channels are required [Aug 18 10:34:33] DEBUG[15372] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15372] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15372] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:33] DEBUG[15372] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15372] bridge.c: Bridge 76c72761-45c7-42a4-9506-6bc68c2e0a5f: calling simple_bridge technology constructor [Aug 18 10:34:33] DEBUG[15372] bridge.c: Bridge 76c72761-45c7-42a4-9506-6bc68c2e0a5f: calling simple_bridge technology start [Aug 18 10:34:33] DEBUG[15361] channel.c: Channel Recorder/ARI-00000053;2 setting write format path: alaw -> slin [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:33] DEBUG[15372] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15372] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.704, detail: [Aug 18 10:34:33] DEBUG[15378] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.704': 0x7f0c300413e0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:801/channel:1629282873.704, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:801/channel:1629282873.704': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:801/channel:1629282873.704, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:801/channel:1629282873.704': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.704, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.704': 0x7f0c300413e0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:52', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-0000003b', '', 'Stasis', 'calls_0', 35, 29, 'ANSWERED', 3, '', '213023', '')] [Aug 18 10:34:33] DEBUG[15379] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15379] http.c: HTTP Request URI is /ari/channels/213043/snoop?app=calls_0&spy=in [Aug 18 10:34:33] DEBUG[15198] res_stasis_recording.c: 1629282870.613: Sending record(213032_vEJKgxiyRBaoxkURsNCEteyerFelGHYd.wav) command [Aug 18 10:34:33] DEBUG[15378] http.c: match request [ari/bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play] with handler [ari] len 3 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (10 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[15380] app.c: play_and_record: , /var/spool/asterisk/recording/213032_vEJKgxiyRBaoxkURsNCEteyerFelGHYd, 'wav' [Aug 18 10:34:33] DEBUG[15380] app.c: Recording Formats: sfmts=wav [Aug 18 10:34:33] VERBOSE[15380] app.c: x=0, open writing: /var/spool/asterisk/recording/213032_vEJKgxiyRBaoxkURsNCEteyerFelGHYd format: wav, 0x7f0c340d9b40 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.705, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.705': 0x7f0c300b36a0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:802/channel:1629282873.705, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:802/channel:1629282873.705': 0x7f0c30006340 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:802/channel:1629282873.705, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:802/channel:1629282873.705': 0x7f0c30006340 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.705, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.705': 0x7f0c300b36a0 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:34:01', '"" <>', '', 's', 'default', 'Snoop/213023-0000000c', 'UnicastRTP/127.0.0.1:50430-0x7f0c7804a920', 'Stasis', 'calls_0', 19, 19, 'ANSWERED', 3, '', '1629282840.199', '')] [Aug 18 10:34:33] DEBUG[15198] http.c: HTTP keeping session open. status_code:201 [Aug 18 10:34:33] DEBUG[15198] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 (Checking To) --From tag as6be1179a --To-tag as34a9f263 [Aug 18 10:34:33] DEBUG[15376] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 686e0cbf1b31d30b1f3c6ab2324650ad@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 474 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #39 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #39)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116832@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 Max-Forwards: 70 From: ;tag=as7025f96b To: Contact: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 58040179 58040179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16686 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116828@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df Max-Forwards: 70 From: ;tag=as7b799277 To: Contact: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 670818967 670818967 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #4 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #4)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116865@178.62.121.41", nonce="328f7607", response="03e2ed289df65658183229073b55646c" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b84ac14 Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116859@178.62.121.41", nonce="4cb8ffaa", response="34d7073f228c83199f1d8a15ca6aeca9" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116827@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d Max-Forwards: 70 From: ;tag=as568913af To: Contact: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 962154366 962154366 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26886e91 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116860@178.62.121.41", nonce="09aacdf1", response="3c5bdea99ab5d5c94251dcfc28700249" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890281 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #119 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #119)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116834@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7aa896bc Max-Forwards: 70 From: ;tag=as2f225ac7 To: Contact: Call-ID: 71babb7e3832ba1353344961206841b0@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 637714792 637714792 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10426 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06fea4c6 Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116856@178.62.121.41", nonce="4ecc103f", response="5984a684e9cf4d2ea38b7a835805bdb8" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029195 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #135 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #135)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116835@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51bde070 Max-Forwards: 70 From: ;tag=as2e8baab5 To: Contact: Call-ID: 533d32874864a5a72dc4aafc53bab0ac@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1705768465 1705768465 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17162 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 From: ;tag=as79336d5f To: ;tag=as2a29680d Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1600602803 1600602803 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 12714 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2304b939;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as79336d5f [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as2a29680d [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1600602803 1600602803 IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 12714 RTP/AVP 0 8 101 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 (Checking To) --From tag as79336d5f --To-tag as2a29680d [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 16541045053beef31ed8e5361337aa22@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 From: ;tag=as10d8c0eb To: ;tag=as70da9059 Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK650b48e7;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as10d8c0eb [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as70da9059 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b9d5690" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 (Checking To) --From tag as10d8c0eb --To-tag as70da9059 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 40cf8f1449337acf099d514511a8313d@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc;received=159.65.48.104 From: ;tag=as0611ab7b To: ;tag=as096aecc1 Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78bb53e7" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK68a34bfc;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0611ab7b [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as096aecc1 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78bb53e7" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 (Checking To) --From tag as0611ab7b --To-tag as096aecc1 [Aug 18 10:34:33] DEBUG[15272] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:33] DEBUG[15378] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15272] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:33] DEBUG[15376] stasis.c: Creating topic. name: bridge:17bb01a4-d2ba-4441-8232-7629d41e0962, detail: [Aug 18 10:34:33] DEBUG[15376] stasis.c: Topic 'bridge:17bb01a4-d2ba-4441-8232-7629d41e0962': 0x7f0c2400d650 created [Aug 18 10:34:33] DEBUG[15376] stasis.c: Creating topic. name: cache:803/bridge:17bb01a4-d2ba-4441-8232-7629d41e0962, detail: [Aug 18 10:34:33] DEBUG[15376] stasis.c: Topic 'cache:803/bridge:17bb01a4-d2ba-4441-8232-7629d41e0962': 0x7f0c24078330 created [Aug 18 10:34:33] DEBUG[15379] http.c: match request [ari/channels/213043/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:33] DEBUG[15272] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:33] DEBUG[15272] channel.c: Channel Announcer/ARI-0000004f;1 setting write format path: slin -> slin [Aug 18 10:34:33] DEBUG[15272] channel.c: Channel 0x7f0c1c048410 'Announcer/ARI-0000004f;1' hanging up. Refs: 2 [Aug 18 10:34:33] DEBUG[15381] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15381] http.c: HTTP Request URI is /ari/bridges [Aug 18 10:34:33] DEBUG[15381] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15381] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15381] http.c: match request [ari/bridges] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15381] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15381] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15381] stasis.c: Creating topic. name: bridge:126b93b8-c878-4567-9b0e-f0027ef4edab, detail: [Aug 18 10:34:33] DEBUG[15381] stasis.c: Topic 'bridge:126b93b8-c878-4567-9b0e-f0027ef4edab': 0x7f0c3014ef90 created [Aug 18 10:34:33] DEBUG[15381] stasis.c: Creating topic. name: cache:804/bridge:126b93b8-c878-4567-9b0e-f0027ef4edab, detail: [Aug 18 10:34:33] DEBUG[15381] stasis.c: Topic 'cache:804/bridge:126b93b8-c878-4567-9b0e-f0027ef4edab': 0x7f0c30162a20 created [Aug 18 10:34:33] DEBUG[15381] bridge_native_rtp.c: Bridge '126b93b8-c878-4567-9b0e-f0027ef4edab' can not use native RTP bridge as two channels are required [Aug 18 10:34:33] DEBUG[15381] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15381] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15381] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:33] DEBUG[15381] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15381] bridge.c: Bridge 126b93b8-c878-4567-9b0e-f0027ef4edab: calling simple_bridge technology constructor [Aug 18 10:34:33] DEBUG[15381] bridge.c: Bridge 126b93b8-c878-4567-9b0e-f0027ef4edab: calling simple_bridge technology start [Aug 18 10:34:33] DEBUG[15379] http.c: match request [ari/channels/213043/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15376] bridge_native_rtp.c: Bridge '17bb01a4-d2ba-4441-8232-7629d41e0962' can not use native RTP bridge as two channels are required [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #22 (4) BYE - 8 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #22)) [Aug 18 10:34:33] VERBOSE[15377] chan_sip.c: Audio is at 15514 [Aug 18 10:34:33] DEBUG[15379] http.c: match request [ari/channels/213043/snoop] with handler [ari] len 3 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: BYE sip:79821117017@178.62.121.41:5060 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK204acf50 Max-Forwards: 70 From: ;tag=as4f7b8e6e To: ;tag=as121e4580 Call-ID: 3aab50745ad148b672e1b26e032bb4fa@159.65.48.104:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821117017@178.62.121.41:5060", nonce="0de4ceb7", response="5f946298dc53c0cdee5695c18ecb459f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'BYE sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15376] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:33] DEBUG[15376] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:33] DEBUG[15376] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Aug 18 10:34:33] DEBUG[15376] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:33] DEBUG[15376] bridge.c: Bridge 17bb01a4-d2ba-4441-8232-7629d41e0962: calling simple_bridge technology constructor [Aug 18 10:34:33] DEBUG[15376] bridge.c: Bridge 17bb01a4-d2ba-4441-8232-7629d41e0962: calling simple_bridge technology start [Aug 18 10:34:33] DEBUG[15376] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15379] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Finding handler for channels/213043/snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Finding handler for 213043 [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channels create: Didn't match 213043 [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channels externalMedia: Didn't match 213043 [Aug 18 10:34:33] DEBUG[15379] res_ari.c: No explicit handler found for 213043. Using wildcard channelId. [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Finding handler for snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:33] DEBUG[15379] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:33] DEBUG[15376] http.c: HTTP closing session. Top level [Aug 18 10:34:33] VERBOSE[15377] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] DEBUG[15383] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:33] VERBOSE[15377] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[15377] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[15383] http.c: HTTP Request URI is /ari/channels/213029/snoop?app=calls_0&spy=in [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Finding handler for bridges/fa4500bd-5aa0-4ef0-bd71-2993d47cd759/play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Finding handler for fa4500bd-5aa0-4ef0-bd71-2993d47cd759 [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15378] res_ari.c: No explicit handler found for fa4500bd-5aa0-4ef0-bd71-2993d47cd759. Using wildcard bridgeId. [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Finding handler for play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:33] DEBUG[15378] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:33] DEBUG[15383] http.c: match request [ari/channels/213029/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15359] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:33] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:33] DEBUG[15383] http.c: match request [ari/channels/213029/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Initializing initreq for method INVITE - callid 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263;received=159.65.48.104 From: ;tag=as7d784780 To: ;tag=as76a7304d Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a209b2b" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[15383] http.c: match request [ari/channels/213029/snoop] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[15384] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15384] http.c: HTTP Request URI is /ari/channels/213032/snoop?app=calls_0&spy=in [Aug 18 10:34:33] DEBUG[15381] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK1b82f263;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d784780 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as76a7304d [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a209b2b" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 (Checking To) --From tag as7d784780 --To-tag as76a7304d [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 1b4e8637043da9713c448cbc29cde2e2@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #89 (3) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #89)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116830@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd Max-Forwards: 70 From: ;tag=as39b021ce To: Contact: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1100223183 1100223183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #118 (1) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #118)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116825@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK608439d6 Max-Forwards: 70 From: ;tag=as5d5338d2 To: Contact: Call-ID: 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 479054008 479054008 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17982 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #117 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #117)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116829@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6da7af93 Max-Forwards: 70 From: ;tag=as5c283596 To: Contact: Call-ID: 1314bee37cb74a327d8c5f1b71c96b89@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2033823795 2033823795 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13902 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #18 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #18)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116862@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2baf6b12 Max-Forwards: 70 From: ;tag=as0bc44772 To: Contact: Call-ID: 0b26716f0c119a15755932c124bb341d@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116862@178.62.121.41", nonce="0b319825", response="6a20eac9692b8e193e543b5bdc77a717" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1645731456 1645731457 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18924 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Destroying SIP dialog 4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '4578951669a241dd1aa3ce322d560ee1@159.65.48.104:5060' Method: BYE [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS stop [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[15383] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116821@178.62.121.41 SIP/2.0 [Aug 18 10:34:33] DEBUG[20585] res_rtp_asterisk.c: (0x7f0cb0015bc0) ICE RTP transport deallocating [Aug 18 10:34:33] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0cb0015bc0' [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 307 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 307 [Aug 18 10:34:33] DEBUG[15381] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15384] http.c: match request [ari/channels/213032/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK104fda02 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Finding handler for channels/213029/snoop [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 From: ;tag=as3d872a68 To: ;tag=as704d0033 Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1966445502 1966445502 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15226 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:33] DEBUG[15384] http.c: match request [ari/channels/213032/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK71fb03b6;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3d872a68 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as704d0033 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1966445502 1966445502 IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15226 RTP/AVP 0 8 101 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:33] DEBUG[14835] res_rtp_asterisk.c: (0x7f0c40036750) RTCP got report of 76 bytes from 178.62.121.41:10973 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:33] DEBUG[15384] http.c: match request [ari/channels/213032/snoop] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15384] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 (Checking To) --From tag as3d872a68 --To-tag as704d0033 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 7dd1b28a0b48543e22864fa05e0991b1@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888;received=159.65.48.104 From: ;tag=as4bc9e76f To: ;tag=as5fdc366e Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="328f7607" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2c65c888;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4bc9e76f [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5fdc366e [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="328f7607" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 (Checking To) --From tag as4bc9e76f --To-tag as5fdc366e [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060' of Request 102: Match Not Found [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 476 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 476 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 458 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 458 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 3 [ 52]: From: ;tag=as3a26bfb9 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Finding handler for channels/213032/snoop [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888;received=159.65.48.104 From: ;tag=as48cc2656 To: ;tag=as6ab88ab6 Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59bb839c" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as48cc2656 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6ab88ab6 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59bb839c" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 (Checking To) --From tag as48cc2656 --To-tag as6ab88ab6 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Finding handler for channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Finding handler for 213032 [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channels create: Didn't match 213032 [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channels externalMedia: Didn't match 213032 [Aug 18 10:34:33] DEBUG[15384] res_ari.c: No explicit handler found for 213032. Using wildcard channelId. [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Finding handler for snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:33] DEBUG[15384] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Finding handler for 213029 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channels create: Didn't match 213029 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channels externalMedia: Didn't match 213029 [Aug 18 10:34:33] DEBUG[15383] res_ari.c: No explicit handler found for 213029. Using wildcard channelId. [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Finding handler for snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:33] DEBUG[15383] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK09d04888 Max-Forwards: 70 From: ;tag=as48cc2656 To: ;tag=as6ab88ab6 Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 6 [ 60]: Call-ID: 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:33 GMT [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] VERBOSE[15377] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116821@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK104fda02 Max-Forwards: 70 From: ;tag=as3a26bfb9 To: Contact: Call-ID: 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1544986695 1544986695 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15514 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #82 [Aug 18 10:34:33] DEBUG[15377] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Audio is at 10056 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e6ce84c Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116863@178.62.121.41", nonce="59bb839c", response="1881b128747b5422f1976fbb90797199" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661388 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #55 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15303] channel.c: Channel 0x7f0c8c11d010 'SIP/zvonobot-000000fc' allocated [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:33] DEBUG[15303] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:33] DEBUG[15303] res_stasis.c: calls_0: Subscribing to 213214 [Aug 18 10:34:33] DEBUG[15303] stasis/app.c: Channel '213214' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15303] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15303] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1;received=159.65.48.104 From: ;tag=as3a399b13 To: ;tag=as3bcd3559 Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68776d7e" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK250b57b1;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as3a399b13 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as3bcd3559 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Outgoing Call for 79821116826 [Aug 18 10:34:33] VERBOSE[15377] dial.c: Called zvonobot/79821116821 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:33] DEBUG[15307] channel.c: Channel 0x7f0c940adc90 'SIP/zvonobot-000000fd' allocated [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:33] DEBUG[15307] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15307] res_stasis.c: calls_0: Subscribing to 213216 [Aug 18 10:34:33] DEBUG[15307] stasis/app.c: Channel '213216' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[15307] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] DEBUG[15307] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Outgoing Call for 79821116824 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68776d7e" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 (Checking To) --From tag as3a399b13 --To-tag as3bcd3559 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 63e4041b488585c57e57de141ed1835f@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #160 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #160)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116872@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7f7a5c57 Max-Forwards: 70 From: ;tag=as23425771 To: Contact: Call-ID: 1e454c24705858e9259d323c756ca026@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116872@178.62.121.41", nonce="35763bfc", response="28335f845f726941e5f5c06d8630a12e" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 20180528 20180529 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12396 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #58 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #58)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116845@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK56c32694 Max-Forwards: 70 From: ;tag=as51a406a7 To: Contact: Call-ID: 27e3ef1421c94d4e6c5e366c5b2b7713@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 82259821 82259821 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] VERBOSE[15386] chan_sip.c: Audio is at 14390 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] VERBOSE[15386] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] VERBOSE[15385] chan_sip.c: Audio is at 15260 [Aug 18 10:34:33] VERBOSE[15386] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[15386] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] VERBOSE[15385] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Initializing initreq for method INVITE - callid 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116824@178.62.121.41 SIP/2.0 [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69fe0e7d [Aug 18 10:34:33] VERBOSE[15385] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 3 [ 52]: From: ;tag=as6f815416 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10;received=159.65.48.104 From: ;tag=as7d725550 To: ;tag=as5f86dce0 Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="153bdf62" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 6 [ 60]: Call-ID: 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 [Aug 18 10:34:33] VERBOSE[15385] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:33 GMT [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Initializing initreq for method INVITE - callid 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116826@178.62.121.41 SIP/2.0 [Aug 18 10:34:33] VERBOSE[15386] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116824@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69fe0e7d Max-Forwards: 70 From: ;tag=as6f815416 To: Contact: Call-ID: 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937031862 937031862 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7881358b [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 3 [ 52]: From: ;tag=as5531004e [Aug 18 10:34:33] DEBUG[15386] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 6 [ 60]: Call-ID: 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:33 GMT [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10;received=159.65.48.104 [Aug 18 10:34:33] VERBOSE[15386] dial.c: Called zvonobot/79821116824 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7d725550 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as5f86dce0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="153bdf62" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] VERBOSE[15385] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116826@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7881358b Max-Forwards: 70 From: ;tag=as5531004e To: Contact: Call-ID: 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 715600489 715600489 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:33] DEBUG[15385] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 (Checking To) --From tag as7d725550 --To-tag as5f86dce0 [Aug 18 10:34:33] DEBUG[13426] chan_sip.c: Hangup call SIP/zvonobot-00000044, SIP callid 72b609e44febd02a23174a4d311879ff@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[13426] res_rtp_asterisk.c: (0x7f0c3c031e30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13426] res_rtp_asterisk.c: (0x7f0c3c031e30) DTLS srtp - stopped timeout timer' [Aug 18 10:34:33] DEBUG[13426] channel.c: Channel 0x7f0c3c04f300 'SIP/zvonobot-00000044' destroying [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 523 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 523 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213028': is 0 interested in calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis/app.c: channel '213028' unsubscribed from calls_0 [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: cache:139/channel:213028, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'cache:139/channel:213028': 0x7f0c3c0513c0 destroyed [Aug 18 10:34:33] DEBUG[20620] stasis.c: Destroying topic. name: channel:213028, detail: [Aug 18 10:34:33] DEBUG[20620] stasis.c: Topic 'channel:213028': 0x7f0c3c0956a0 destroyed [Aug 18 10:34:33] DEBUG[15310] channel.c: Channel 0x7f0c90072ad0 'SIP/zvonobot-000000fe' allocated [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:33] DEBUG[15310] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:33] DEBUG[15310] res_stasis.c: calls_0: Subscribing to 213217 [Aug 18 10:34:33] DEBUG[15310] stasis/app.c: Channel '213217' is 1 interested in calls_0 [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: channel:1629282873.706, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.706': 0x7f0c3006c950 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Creating topic. name: cache:805/channel:1629282873.706, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:805/channel:1629282873.706': 0x7f0c300fcee0 created [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: cache:805/channel:1629282873.706, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'cache:805/channel:1629282873.706': 0x7f0c300fcee0 destroyed [Aug 18 10:34:33] DEBUG[20545] stasis.c: Destroying topic. name: channel:1629282873.706, detail: [Aug 18 10:34:33] DEBUG[20545] stasis.c: Topic 'channel:1629282873.706': 0x7f0c3006c950 destroyed [Aug 18 10:34:33] DEBUG[20545] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate", "clid", "src", "dst", "dcontext", "channel", "dstchannel", "lastapp", "lastdata", "duration", "billsec", "disposition", "amaflags", "accountcode", "uniqueid", "userfield") VALUES ('2021-08-18 10:33:54', '"" <74950493843>', '74950493843', '', 'incoming-call', 'SIP/zvonobot-00000044', '', 'Stasis', 'calls_0', 37, 5, 'ANSWERED', 3, '', '213028', '')] [Aug 18 10:34:33] DEBUG[15064] channel.c: Channel 0x7f0c780710e0 'Snoop/212969-0000001f' allocated [Aug 18 10:34:33] DEBUG[15064] channel.c: Channel 0x7f0c780710e0 'Snoop/212969-0000001f' hanging up. Refs: 3 [Aug 18 10:34:33] DEBUG[15064] autoservice.c: Thread is a user interface, not removing channel Snoop/212969-0000001f from autoservice [Aug 18 10:34:33] DEBUG[15310] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:33] VERBOSE[15385] dial.c: Called zvonobot/79821116826 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Outgoing Call for 79821116823 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] VERBOSE[15387] chan_sip.c: Audio is at 11192 [Aug 18 10:34:33] VERBOSE[15387] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] DEBUG[15310] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:33] VERBOSE[15387] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[15387] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Initializing initreq for method INVITE - callid 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116823@178.62.121.41 SIP/2.0 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a639eb [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 3 [ 52]: From: ;tag=as426458c9 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 6 [ 60]: Call-ID: 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:33 GMT [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] VERBOSE[15387] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116823@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a639eb Max-Forwards: 70 From: ;tag=as426458c9 To: Contact: Call-ID: 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1198383622 1198383622 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #114 [Aug 18 10:34:33] DEBUG[15387] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Aug 18 10:34:33] VERBOSE[15387] dial.c: Called zvonobot/79821116823 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:33] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '3334457d359183f4783a1ff53f046a25@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK264abc10 Max-Forwards: 70 From: ;tag=as7d725550 To: ;tag=as5f86dce0 Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Audio is at 16168 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04faa4b4 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116867@178.62.121.41", nonce="153bdf62", response="88c633d268e35d5ca7cd4960941d4e3b" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706902 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #118 (2) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #118)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116825@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK608439d6 Max-Forwards: 70 From: ;tag=as5d5338d2 To: Contact: Call-ID: 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 479054008 479054008 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17982 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #99 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #99)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116871@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK21c7db71 Max-Forwards: 70 From: ;tag=as0611ab7b To: Contact: Call-ID: 7cb61d7209c0c3400b948bea346ff7d0@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116871@178.62.121.41", nonce="78bb53e7", response="9c83b53faaf9a44a25c9afd6a956a8cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1679218160 1679218161 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12786 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #25 (4) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #25)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116833@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK77976a1d Max-Forwards: 70 From: ;tag=as235fcd7f To: Contact: Call-ID: 4903609f55c0afa865ee7d465f9e75a2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1532262111 1532262111 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14362 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #76 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #76)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116846@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26314b07 Max-Forwards: 70 From: ;tag=as2a65fe21 To: Contact: Call-ID: 6a7c51965ceaaa5145e1888e5911ba1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 513276323 513276323 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19816 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #82 (1) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #82)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116821@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK104fda02 Max-Forwards: 70 From: ;tag=as3a26bfb9 To: Contact: Call-ID: 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1544986695 1544986695 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15514 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (1) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e6ce84c Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116863@178.62.121.41", nonce="59bb839c", response="1881b128747b5422f1976fbb90797199" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661388 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (5) INVITE - 5 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #45)) [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116874@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4c8f58 Max-Forwards: 70 From: ;tag=as3f6b0566 To: Contact: Call-ID: 7824d5b579c0c91b0060a71a25625e56@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116874@178.62.121.41", nonce="6dec0191", response="9de207a8298805167b35037460b5a7bc" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 970235664 970235665 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 From: ;tag=as57de5f2f To: ;tag=as7e2e6628 Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 CSeq: 103 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 266 v=0 o=root 1817551672 1817551672 IN IP4 178.62.121.41 s=Asterisk PBX 16.18.0 c=IN IP4 178.62.121.41 t=0 0 m=audio 15352 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK51b61fcf;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as57de5f2f [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7e2e6628 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 45]: Contact: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 12 [ 14]: Require: timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 13 [ 19]: Content-Length: 266 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 14 [ 0]: [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 0 [ 3]: v=0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 1 [ 49]: o=root 1817551672 1817551672 IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 3 [ 22]: c=IN IP4 178.62.121.41 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 5 [ 29]: m=audio 15352 RTP/AVP 0 8 101 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 10 [ 14]: a=maxptime:150 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (14 headers 12 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 (Checking To) --From tag as57de5f2f --To-tag as7e2e6628 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 596ef69a675656833620d8345b47f33c@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 827 [Aug 18 10:34:33] VERBOSE[20530] asterisk.c: Remote UNIX connection [Aug 18 10:34:33] DEBUG[14742] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:33] VERBOSE[15394] asterisk.c: Remote UNIX connection disconnected [Aug 18 10:34:33] DEBUG[15214] http.c: HTTP keeping session open. status_code:404 [Aug 18 10:34:33] DEBUG[15214] http.c: HTTP closing session. Top level [Aug 18 10:34:33] DEBUG[15398] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[15398] http.c: HTTP Request URI is /ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/play?media=sound%3Asilence%2F2 [Aug 18 10:34:33] DEBUG[15398] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/play] with handler [httpstatus] len 10 [Aug 18 10:34:33] DEBUG[15398] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/play] with handler [phoneprov] len 9 [Aug 18 10:34:33] DEBUG[15398] http.c: match request [ari/bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/play] with handler [ari] len 3 [Aug 18 10:34:33] DEBUG[15398] http.c: Match made with [ari] [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Finding handler for bridges/f58763a3-c201-4609-b9b6-f8cb14b257ad/play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Finding handler for bridges [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0;received=159.65.48.104 From: ;tag=as4b888052 To: ;tag=as7344fb7b Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71b13d8e" Content-Length: 0 <-------------> [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0;received=159.65.48.104 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as4b888052 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as7344fb7b [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71b13d8e" [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:33] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: = Looking for Call ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 (Checking To) --From tag as4b888052 --To-tag as7344fb7b [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari endpoints: Didn't match bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari events: Didn't match bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari applications: Didn't match bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari recordings: Didn't match bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari channels: Didn't match bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking ari bridges: Explicit match with bridges [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Finding handler for f58763a3-c201-4609-b9b6-f8cb14b257ad [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridges bridgeId: Matched wildcard. [Aug 18 10:34:33] DEBUG[15398] res_ari.c: No explicit handler found for f58763a3-c201-4609-b9b6-f8cb14b257ad. Using wildcard bridgeId. [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Finding handler for play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridgeId addChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridgeId removeChannel: Didn't match play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridgeId videoSource: Didn't match play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridgeId moh: Didn't match play [Aug 18 10:34:33] DEBUG[15398] res_ari.c: Checking bridgeId play: Explicit match with play [Aug 18 10:34:33] DEBUG[15400] http.c: HTTP opening session. Top level [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #142 [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: Stopping retransmission on '40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:33] DEBUG[15398] stasis.c: Creating topic. name: channel:1629282873.707, detail: [Aug 18 10:34:33] DEBUG[15398] stasis.c: Topic 'channel:1629282873.707': 0x7f0c7804ece0 created [Aug 18 10:34:33] DEBUG[15398] stasis.c: Creating topic. name: cache:806/channel:1629282873.707, detail: [Aug 18 10:34:33] DEBUG[15398] stasis.c: Topic 'cache:806/channel:1629282873.707': 0x7f0c78029310 created [Aug 18 10:34:33] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:33] DEBUG[15400] http.c: HTTP Request URI is /ari/channels/externalMedia?channelId=robot_213016&app=calls_0&format=slin16&external_host=127.0.0.1%3A50395 [Aug 18 10:34:33] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67c6bcf0 Max-Forwards: 70 From: ;tag=as4b888052 To: ;tag=as7344fb7b Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Audio is at 17348 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5532a489 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116854@178.62.121.41", nonce="71b13d8e", response="1085dd2b15e8d0125cea2557194bb890" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090438 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #123 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15400] http.c: match request [ari/channels/externalMedia] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15319] channel.c: Channel 0x7f0ca4060120 'SIP/zvonobot-00000101' allocated [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:34] DEBUG[15319] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #23 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #23)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116839@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK67baf99f Max-Forwards: 70 From: ;tag=as3c16b086 To: Contact: Call-ID: 307304ec558a22464cac7a9262f07b46@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 213170282 213170282 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15154 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116824@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69fe0e7d Max-Forwards: 70 From: ;tag=as6f815416 To: Contact: Call-ID: 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937031862 937031862 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116826@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7881358b Max-Forwards: 70 From: ;tag=as5531004e To: Contact: Call-ID: 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 715600489 715600489 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #125 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #125)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116875@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK549550f0 Max-Forwards: 70 From: ;tag=as7e29cf80 To: Contact: Call-ID: 7c6bef2b5745d32343661d86626fbc48@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116875@178.62.121.41", nonce="307fff96", response="33cb57553fa8063f9e0b9e5897d29958" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1196754311 1196754312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #50 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #50)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116877@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK63e0761e Max-Forwards: 70 From: ;tag=as06a66f45 To: Contact: Call-ID: 062e901479878f3469dc381c0f75eb83@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116877@178.62.121.41", nonce="749a75da", response="2298fba5ba0d456fa3928840d0b1f2cd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 187336558 187336559 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16822 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #44 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #44)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116873@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK20c42335 Max-Forwards: 70 From: ;tag=as109bf1f8 To: Contact: Call-ID: 6a9045fb1905dd407eab47186c096641@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116873@178.62.121.41", nonce="489df732", response="cb32645ad2f010aeeec7afb1f22a36bd" Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 199028039 199028040 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13196 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15400] http.c: match request [ari/channels/externalMedia] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15411] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15411] http.c: HTTP Request URI is /ari/channels/213224?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116816&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15411] http.c: match request [ari/channels/213224] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15411] http.c: match request [ari/channels/213224] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15411] http.c: match request [ari/channels/213224] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15411] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15411] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15400] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Finding handler for channels/213224 [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15400] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Finding handler for channels/externalMedia [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Finding handler for 213224 [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking channels create: Didn't match 213224 [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15411] res_ari.c: Checking channels externalMedia: Didn't match 213224 [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15315] channel.c: Channel 0x7f0ca8007d50 'SIP/zvonobot-000000ff' allocated [Aug 18 10:34:34] DEBUG[15411] res_ari.c: No explicit handler found for 213224. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:34] DEBUG[15315] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15413] http.c: HTTP opening session. Top level [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98;received=159.65.48.104 From: ;tag=as5f8e1c47 To: ;tag=as6c0630b9 Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71de2929" Content-Length: 0 <-------------> [Aug 18 10:34:34] DEBUG[15413] http.c: HTTP Request URI is /ari/channels/213225?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116815&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15413] http.c: match request [ari/channels/213225] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Finding handler for externalMedia [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking channels create: Didn't match externalMedia [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15400] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [Aug 18 10:34:34] DEBUG[15400] netsock2.c: Splitting '127.0.0.1:50395' into... [Aug 18 10:34:34] DEBUG[15400] netsock2.c: ...host '127.0.0.1' and port '50395'. [Aug 18 10:34:34] DEBUG[15400] netsock2.c: Splitting '127.0.0.1:50395' into... [Aug 18 10:34:34] DEBUG[15400] netsock2.c: ...host '127.0.0.1' and port '50395'. [Aug 18 10:34:34] DEBUG[15400] acl.c: For destination '127.0.0.1', our source address is '127.0.0.1'. [Aug 18 10:34:34] DEBUG[15400] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c841544d0' [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) RTP allocated port 11814 [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) ICE creating session 127.0.0.1:11814 (11814) [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) ICE create [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) ICE add system candidates [Aug 18 10:34:34] DEBUG[15400] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15400] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15223] stasis.c: Creating topic. name: channel:1629282874.708, detail: [Aug 18 10:34:34] DEBUG[15223] stasis.c: Topic 'channel:1629282874.708': 0x7f0cb00b15f0 created [Aug 18 10:34:34] DEBUG[15223] stasis.c: Creating topic. name: cache:807/channel:1629282874.708, detail: [Aug 18 10:34:34] DEBUG[15413] http.c: match request [ari/channels/213225] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:34] DEBUG[15223] stasis.c: Topic 'cache:807/channel:1629282874.708': 0x7f0cb005bc10 created [Aug 18 10:34:34] DEBUG[15413] http.c: match request [ari/channels/213225] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15315] res_stasis.c: calls_0: Subscribing to 213218 [Aug 18 10:34:34] DEBUG[15315] stasis/app.c: Channel '213218' is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98;received=159.65.48.104 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as5f8e1c47 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as6c0630b9 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71de2929" [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: = Looking for Call ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 (Checking To) --From tag as5f8e1c47 --To-tag as6c0630b9 [Aug 18 10:34:34] DEBUG[15413] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15413] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Finding handler for channels/213225 [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Finding handler for 213225 [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking channels create: Didn't match 213225 [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15413] res_ari.c: Checking channels externalMedia: Didn't match 213225 [Aug 18 10:34:34] DEBUG[15413] res_ari.c: No explicit handler found for 213225. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15319] res_stasis.c: calls_0: Subscribing to 213222 [Aug 18 10:34:34] DEBUG[15415] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15315] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:34] DEBUG[15315] http.c: HTTP closing session. Top level [Aug 18 10:34:34] DEBUG[15423] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15319] stasis/app.c: Channel '213222' is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[15415] http.c: HTTP Request URI is /ari/channels/213226?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116814&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Outgoing Call for 79821116818 [Aug 18 10:34:34] DEBUG[15415] http.c: match request [ari/channels/213226] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15415] http.c: match request [ari/channels/213226] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: Allocating new SIP dialog for 74c0b853516f3b78435181b6592cd75f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15411] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8009fc00' [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) RTP allocated port 10518 [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE creating session 0.0.0.0:10518 (10518) [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE create [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE add system candidates [Aug 18 10:34:34] DEBUG[15411] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15411] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE add candidate: 159.65.48.104:10518, 2130706431 [Aug 18 10:34:34] DEBUG[15411] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15411] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE add candidate: 10.131.0.10:10518, 2130706431 [Aug 18 10:34:34] DEBUG[15411] rtp_engine.c: RTP instance '0x7f0c8009fc00' is setup and ready to go [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) ICE stopped [Aug 18 10:34:34] DEBUG[15319] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Acked pending invite 102 [Aug 18 10:34:34] DEBUG[15415] http.c: match request [ari/channels/213226] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15373] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:34] DEBUG[15319] http.c: HTTP closing session. Top level [Aug 18 10:34:34] DEBUG[15426] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) ICE add candidate: 159.65.48.104:11814, 2130706431 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #86 [Aug 18 10:34:34] DEBUG[15415] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15415] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Finding handler for channels/213226 [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Finding handler for 213226 [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking channels create: Didn't match 213226 [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15415] res_ari.c: Checking channels externalMedia: Didn't match 213226 [Aug 18 10:34:34] DEBUG[15415] res_ari.c: No explicit handler found for 213226. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15273] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15426] http.c: HTTP Request URI is /ari/channels/213230?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116810&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15263] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:34] DEBUG[15411] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15426] http.c: match request [ari/channels/213230] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15426] http.c: match request [ari/channels/213230] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Stopping retransmission on '21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060' of Request 102: Match Found [Aug 18 10:34:34] DEBUG[15423] http.c: HTTP Request URI is /ari/channels/213229?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116811&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP response 401 to standard invite [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Transmitting (no NAT) to 178.62.121.41:5060: ACK sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK213ccc98 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: ;tag=as6c0630b9 Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.20.0 Content-Length: 0 --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'ACK sip:798' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Auth attempt 1 on INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Audio is at 19388 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Outgoing Call for 79821116822 [Aug 18 10:34:34] DEBUG[15400] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15400] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15400] res_rtp_asterisk.c: (0x7f0c841544d0) ICE add candidate: 10.131.0.10:11814, 2130706431 [Aug 18 10:34:34] DEBUG[15400] rtp_engine.c: RTP instance '0x7f0c841544d0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15400] stasis.c: Creating topic. name: channel:robot_213016, detail: [Aug 18 10:34:34] DEBUG[15400] stasis.c: Topic 'channel:robot_213016': 0x7f0c84141d40 created [Aug 18 10:34:34] DEBUG[15400] stasis.c: Creating topic. name: cache:808/channel:robot_213016, detail: [Aug 18 10:34:34] DEBUG[15400] stasis.c: Topic 'cache:808/channel:robot_213016': 0x7f0c84099e40 created [Aug 18 10:34:34] DEBUG[15423] http.c: match request [ari/channels/213229] with handler [httpstatus] len 10 [Aug 18 10:34:34] VERBOSE[15424] chan_sip.c: Audio is at 18248 [Aug 18 10:34:34] DEBUG[15423] http.c: match request [ari/channels/213229] with handler [phoneprov] len 9 [Aug 18 10:34:34] VERBOSE[15424] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] VERBOSE[15424] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] VERBOSE[15424] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Initializing initreq for method INVITE - callid 359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116818@178.62.121.41 SIP/2.0 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4f86ec [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 3 [ 52]: From: ;tag=as0c05fdf2 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 6 [ 60]: Call-ID: 359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:34 GMT [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:34] VERBOSE[15424] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116818@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4f86ec Max-Forwards: 70 From: ;tag=as0c05fdf2 To: Contact: Call-ID: 359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 161411965 161411965 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18248 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Aug 18 10:34:34] DEBUG[15424] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:34] DEBUG[15423] http.c: match request [ari/channels/213229] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15426] http.c: match request [ari/channels/213230] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15411] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15411] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15411] res_rtp_asterisk.c: (0x7f0c8009fc00) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15411] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15411] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15411] chan_sip.c: SIP call-id changed from '74c0b853516f3b78435181b6592cd75f@127.0.1.1:5060' to '691402880702a202113696990b3a377f@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15411] stasis.c: Creating topic. name: channel:213224, detail: [Aug 18 10:34:34] DEBUG[15411] stasis.c: Topic 'channel:213224': 0x7f0c80029250 created [Aug 18 10:34:34] DEBUG[15411] stasis.c: Creating topic. name: cache:809/channel:213224, detail: [Aug 18 10:34:34] DEBUG[15411] stasis.c: Topic 'cache:809/channel:213224': 0x7f0c80128af0 created [Aug 18 10:34:34] VERBOSE[15424] dial.c: Called zvonobot/79821116818 [Aug 18 10:34:34] DEBUG[15427] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: Allocating new SIP dialog for 1b4e3f1e0c40213575a0178a58f4452d@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15413] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8c06cac0' [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) RTP allocated port 16052 [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE creating session 0.0.0.0:16052 (16052) [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE create [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE add system candidates [Aug 18 10:34:34] DEBUG[15413] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15413] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE add candidate: 159.65.48.104:16052, 2130706431 [Aug 18 10:34:34] DEBUG[15413] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15413] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE add candidate: 10.131.0.10:16052, 2130706431 [Aug 18 10:34:34] DEBUG[15413] rtp_engine.c: RTP instance '0x7f0c8c06cac0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) ICE stopped [Aug 18 10:34:34] DEBUG[15413] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15413] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15413] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15413] res_rtp_asterisk.c: (0x7f0c8c06cac0) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15413] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15413] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15413] chan_sip.c: SIP call-id changed from '1b4e3f1e0c40213575a0178a58f4452d@127.0.1.1:5060' to '772029a2578ad83d31d443d1014e93b4@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15413] stasis.c: Creating topic. name: channel:213225, detail: [Aug 18 10:34:34] DEBUG[15413] stasis.c: Topic 'channel:213225': 0x7f0c8c0498e0 created [Aug 18 10:34:34] DEBUG[15413] stasis.c: Creating topic. name: cache:810/channel:213225, detail: [Aug 18 10:34:34] DEBUG[15413] stasis.c: Topic 'cache:810/channel:213225': 0x7f0c8c058b00 created [Aug 18 10:34:34] DEBUG[15273] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:34] DEBUG[15273] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:34] DEBUG[15273] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 18 10:34:34] DEBUG[15273] channel.c: Channel Announcer/ARI-0000004e;1 setting write format path: slin -> slin [Aug 18 10:34:34] DEBUG[15423] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15423] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Finding handler for channels/213229 [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Finding handler for 213229 [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking channels create: Didn't match 213229 [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15423] res_ari.c: Checking channels externalMedia: Didn't match 213229 [Aug 18 10:34:34] DEBUG[15423] res_ari.c: No explicit handler found for 213229. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15426] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 305 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 305 [Aug 18 10:34:34] DEBUG[15273] channel.c: Channel 0x7f0c1805f5c0 'Announcer/ARI-0000004e;1' hanging up. Refs: 2 [Aug 18 10:34:34] DEBUG[15427] http.c: HTTP Request URI is /ari/channels/213227?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116813&callerId=74950493843 [Aug 18 10:34:34] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15427] http.c: match request [ari/channels/213227] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15428] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15427] http.c: match request [ari/channels/213227] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15426] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15427] http.c: match request [ari/channels/213227] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15427] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15427] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Finding handler for channels/213227 [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Finding handler for 213227 [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking channels create: Didn't match 213227 [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15427] res_ari.c: Checking channels externalMedia: Didn't match 213227 [Aug 18 10:34:34] DEBUG[15427] res_ari.c: No explicit handler found for 213227. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Finding handler for channels/213230 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15429] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15429] http.c: HTTP Request URI is /ari/channels/213232?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116808&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15428] http.c: HTTP Request URI is /ari/channels/213231?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116809&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] VERBOSE[15416] chan_sip.c: Audio is at 14728 [Aug 18 10:34:34] DEBUG[15429] http.c: match request [ari/channels/213232] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15429] http.c: match request [ari/channels/213232] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15429] http.c: match request [ari/channels/213232] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15429] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15429] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Finding handler for channels/213232 [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ecf3464 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116851@178.62.121.41", nonce="71de2929", response="a192cb8c01ae51fd5bd60899bae739de" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876616 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116823@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a639eb Max-Forwards: 70 From: ;tag=as426458c9 To: Contact: Call-ID: 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1198383622 1198383622 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Finding handler for channels [Aug 18 10:34:34] VERBOSE[15416] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Finding handler for 213230 [Aug 18 10:34:34] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking channels create: Didn't match 213230 [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15430] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: Allocating new SIP dialog for 49a42fb2431bcc0c6353bfa1595fe8f9@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15415] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c8807e3f0' [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) RTP allocated port 15448 [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE creating session 0.0.0.0:15448 (15448) [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE create [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE add system candidates [Aug 18 10:34:34] DEBUG[15415] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15415] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE add candidate: 159.65.48.104:15448, 2130706431 [Aug 18 10:34:34] DEBUG[15415] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15415] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE add candidate: 10.131.0.10:15448, 2130706431 [Aug 18 10:34:34] DEBUG[15415] rtp_engine.c: RTP instance '0x7f0c8807e3f0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) ICE stopped [Aug 18 10:34:34] DEBUG[15415] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15415] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15415] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15415] res_rtp_asterisk.c: (0x7f0c8807e3f0) RTCP setup on RTP instance [Aug 18 10:34:34] DEBUG[15430] http.c: HTTP Request URI is /ari/channels/213228?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116812&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15426] res_ari.c: Checking channels externalMedia: Didn't match 213230 [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15430] http.c: match request [ari/channels/213228] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15430] http.c: match request [ari/channels/213228] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15430] http.c: match request [ari/channels/213228] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15430] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15426] res_ari.c: No explicit handler found for 213230. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #69 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #69)) [Aug 18 10:34:34] DEBUG[15430] http.c: HTTP consuming request body [Aug 18 10:34:34] VERBOSE[15416] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116843@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK133b00a1 Max-Forwards: 70 From: ;tag=as64111725 To: Contact: Call-ID: 136506452fb77d015d9a8b8e03acd1eb@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1813558790 1813558790 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14230 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Finding handler for channels/213228 [Aug 18 10:34:34] DEBUG[15428] http.c: match request [ari/channels/213231] with handler [httpstatus] len 10 [Aug 18 10:34:34] VERBOSE[15415] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15415] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15415] chan_sip.c: SIP call-id changed from '49a42fb2431bcc0c6353bfa1595fe8f9@127.0.1.1:5060' to '44a425a82ac69af90c9826a12f4862bb@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15415] stasis.c: Creating topic. name: channel:213226, detail: [Aug 18 10:34:34] DEBUG[15415] stasis.c: Topic 'channel:213226': 0x7f0c880843f0 created [Aug 18 10:34:34] DEBUG[15415] stasis.c: Creating topic. name: cache:811/channel:213226, detail: [Aug 18 10:34:34] DEBUG[15415] stasis.c: Topic 'cache:811/channel:213226': 0x7f0c880785a0 created [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15428] http.c: match request [ari/channels/213231] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Finding handler for 213232 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[15431] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15431] http.c: HTTP Request URI is /ari/channels/213233?app=calls_0&timeout=60&endpoint=SIP%2Fzvonobot%2F79821116807&callerId=74950493843 [Aug 18 10:34:34] DEBUG[15428] http.c: match request [ari/channels/213231] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking channels create: Didn't match 213232 [Aug 18 10:34:34] DEBUG[15321] channel.c: Channel 0x7f0cac02d090 'SIP/zvonobot-00000100' allocated [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:34] DEBUG[15321] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15429] res_ari.c: Checking channels externalMedia: Didn't match 213232 [Aug 18 10:34:34] DEBUG[15429] res_ari.c: No explicit handler found for 213232. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:34] DEBUG[15428] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15431] http.c: match request [ari/channels/213233] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: Allocating new SIP dialog for 2bbc27bd2aea60f930a59a5f3271450a@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15427] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c98042000' [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) RTP allocated port 19114 [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE creating session 0.0.0.0:19114 (19114) [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE create [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE add system candidates [Aug 18 10:34:34] DEBUG[15427] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15427] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE add candidate: 159.65.48.104:19114, 2130706431 [Aug 18 10:34:34] DEBUG[15427] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15427] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE add candidate: 10.131.0.10:19114, 2130706431 [Aug 18 10:34:34] DEBUG[15427] rtp_engine.c: RTP instance '0x7f0c98042000' is setup and ready to go [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) ICE stopped [Aug 18 10:34:34] DEBUG[15427] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15427] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15427] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15427] res_rtp_asterisk.c: (0x7f0c98042000) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15427] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15427] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15427] chan_sip.c: SIP call-id changed from '2bbc27bd2aea60f930a59a5f3271450a@127.0.1.1:5060' to '27b060f81f95c1bd0ddf3711418ca59a@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15427] stasis.c: Creating topic. name: channel:213227, detail: [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[14015] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15431] http.c: match request [ari/channels/213233] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15428] http.c: HTTP consuming request body [Aug 18 10:34:34] DEBUG[15431] http.c: match request [ari/channels/213233] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15427] stasis.c: Topic 'channel:213227': 0x7f0c9805b490 created [Aug 18 10:34:34] DEBUG[15431] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15427] stasis.c: Creating topic. name: cache:812/channel:213227, detail: [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Finding handler for channels/213231 [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Finding handler for 213231 [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking channels create: Didn't match 213231 [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15428] res_ari.c: Checking channels externalMedia: Didn't match 213231 [Aug 18 10:34:34] DEBUG[15428] res_ari.c: No explicit handler found for 213231. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15431] http.c: HTTP consuming request body [Aug 18 10:34:34] VERBOSE[15416] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Finding handler for 213228 [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking channels create: Didn't match 213228 [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Finding handler for channels/213233 [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Finding handler for channels [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04faa4b4 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116867@178.62.121.41", nonce="153bdf62", response="88c633d268e35d5ca7cd4960941d4e3b" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706902 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #43 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #43)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116831@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK22357d69 Max-Forwards: 70 From: ;tag=as655f7a70 To: Contact: Call-ID: 7530856d32712b46529332693455f8c4@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2053228906 2053228906 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19488 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #152 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #152)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116838@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK40cd8354 Max-Forwards: 70 From: ;tag=as768f4f04 To: Contact: Call-ID: 104c5130341091f56623cd02618893c9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1482982810 1482982810 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12766 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #82 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[15430] res_ari.c: Checking channels externalMedia: Didn't match 213228 [Aug 18 10:34:34] DEBUG[15427] stasis.c: Topic 'cache:812/channel:213227': 0x7f0c9805c0d0 created [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #82)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116821@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK104fda02 Max-Forwards: 70 From: ;tag=as3a26bfb9 To: Contact: Call-ID: 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1544986695 1544986695 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15514 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15430] res_ari.c: No explicit handler found for 213228. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e6ce84c Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116863@178.62.121.41", nonce="59bb839c", response="1881b128747b5422f1976fbb90797199" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661388 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Finding handler for 213233 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #67 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking channels create: Didn't match 213233 [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #67)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116841@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7708199a Max-Forwards: 70 From: ;tag=as70796578 To: Contact: Call-ID: 6a9b837a2988cd8b57b50c5d1f712748@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1553573500 1553573500 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15450 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #72 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #72)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116840@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK353c2a4a Max-Forwards: 70 From: ;tag=as6ca8b41e To: Contact: Call-ID: 6e35e411133b96ef093204ef3a02be07@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1673363128 1673363128 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10424 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5532a489 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116854@178.62.121.41", nonce="71b13d8e", response="1085dd2b15e8d0125cea2557194bb890" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090438 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116824@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69fe0e7d Max-Forwards: 70 From: ;tag=as6f815416 To: Contact: Call-ID: 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937031862 937031862 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116826@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7881358b Max-Forwards: 70 From: ;tag=as5531004e To: Contact: Call-ID: 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 715600489 715600489 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: Allocating new SIP dialog for 662a8a8256af8fe931dc9111131388a7@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15423] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c90019380' [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) RTP allocated port 13068 [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE creating session 0.0.0.0:13068 (13068) [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE create [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE add system candidates [Aug 18 10:34:34] DEBUG[15423] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15423] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE add candidate: 159.65.48.104:13068, 2130706431 [Aug 18 10:34:34] DEBUG[15423] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15423] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE add candidate: 10.131.0.10:13068, 2130706431 [Aug 18 10:34:34] DEBUG[15423] rtp_engine.c: RTP instance '0x7f0c90019380' is setup and ready to go [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) ICE stopped [Aug 18 10:34:34] DEBUG[15423] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15423] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15423] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15423] res_rtp_asterisk.c: (0x7f0c90019380) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15423] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #96 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #96)) [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Initializing initreq for method INVITE - callid 509f8e0e3acc6f0a7d623d30256c2b1f@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15423] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15431] res_ari.c: Checking channels externalMedia: Didn't match 213233 [Aug 18 10:34:34] DEBUG[15431] res_ari.c: No explicit handler found for 213233. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116822@178.62.121.41 SIP/2.0 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116837@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK290a6082 Max-Forwards: 70 From: ;tag=as6760f146 To: Contact: Call-ID: 4aba017f001e794772eb22a559a13c7f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 612919414 612919414 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12398 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c492a07 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: Allocating new SIP dialog for 7b0828162b5212fa67b0c0546920e4ba@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 From: ;tag=as7eb98fd0 To: ;tag=as30588395 Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" Content-Length: 0 <-------------> [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0df8fcea;received=159.65.48.104 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as7eb98fd0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as30588395 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 3 [ 52]: From: ;tag=as1abebfe3 [Aug 18 10:34:34] DEBUG[15321] res_stasis.c: calls_0: Subscribing to 213223 [Aug 18 10:34:34] DEBUG[15321] stasis/app.c: Channel '213223' is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[15321] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:34] DEBUG[15429] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca4070a50' [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) RTP allocated port 11868 [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2547441d" [Aug 18 10:34:34] DEBUG[15321] http.c: HTTP closing session. Top level [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 6 [ 60]: Call-ID: 509f8e0e3acc6f0a7d623d30256c2b1f@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15423] chan_sip.c: SIP call-id changed from '662a8a8256af8fe931dc9111131388a7@127.0.1.1:5060' to '32816c3d3e1f927a1d65c7db6472a348@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15423] stasis.c: Creating topic. name: channel:213229, detail: [Aug 18 10:34:34] DEBUG[15423] stasis.c: Topic 'channel:213229': 0x7f0c90008040 created [Aug 18 10:34:34] DEBUG[15423] stasis.c: Creating topic. name: cache:813/channel:213229, detail: [Aug 18 10:34:34] DEBUG[15423] stasis.c: Topic 'cache:813/channel:213229': 0x7f0c90058bc0 created [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE creating session 0.0.0.0:11868 (11868) [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Outgoing Call for 79821116817 [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE create [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE add system candidates [Aug 18 10:34:34] DEBUG[15429] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15429] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE add candidate: 159.65.48.104:11868, 2130706431 [Aug 18 10:34:34] DEBUG[15429] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:34] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:34] DEBUG[15429] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE add candidate: 10.131.0.10:11868, 2130706431 [Aug 18 10:34:34] DEBUG[15429] rtp_engine.c: RTP instance '0x7f0ca4070a50' is setup and ready to go [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: = Looking for Call ID: 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 (Checking To) --From tag as7eb98fd0 --To-tag as30588395 [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) ICE stopped [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:34 GMT [Aug 18 10:34:34] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[14533] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:34] VERBOSE[15416] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116822@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c492a07 Max-Forwards: 70 From: ;tag=as1abebfe3 To: Contact: Call-ID: 509f8e0e3acc6f0a7d623d30256c2b1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 812289418 812289418 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14728 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Aug 18 10:34:34] DEBUG[15429] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15429] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 42462bcf58720fbb2059b6de455547db@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15429] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15429] res_rtp_asterisk.c: (0x7f0ca4070a50) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15429] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15429] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15429] chan_sip.c: SIP call-id changed from '7b0828162b5212fa67b0c0546920e4ba@127.0.1.1:5060' to '5c39479065c3f3a73e54d79b54a84f7d@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15429] stasis.c: Creating topic. name: channel:213232, detail: [Aug 18 10:34:34] DEBUG[15429] stasis.c: Topic 'channel:213232': 0x7f0ca4126e70 created [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: Allocating new SIP dialog for 5a9ed7924f0f61bc7d006fd0291f6640@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15430] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cb00883d0' [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) RTP allocated port 16724 [Aug 18 10:34:34] DEBUG[15429] stasis.c: Creating topic. name: cache:814/channel:213232, detail: [Aug 18 10:34:34] DEBUG[15429] stasis.c: Topic 'cache:814/channel:213232': 0x7f0ca4137710 created [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE creating session 0.0.0.0:16724 (16724) [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE create [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE add system candidates [Aug 18 10:34:34] DEBUG[15416] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: Allocating new SIP dialog for 203738434c61173b0bf7cf34436fa76f@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15426] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0c9c0b18f0' [Aug 18 10:34:34] DEBUG[14737] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #39 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #39)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116832@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK578cdbe1 Max-Forwards: 70 From: ;tag=as7025f96b To: Contact: Call-ID: 31f6c3a516708506588cba5558cae631@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 58040179 58040179 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16686 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15430] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) RTP allocated port 16544 [Aug 18 10:34:34] DEBUG[15430] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE add candidate: 159.65.48.104:16724, 2130706431 [Aug 18 10:34:34] DEBUG[15430] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15430] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE add candidate: 10.131.0.10:16724, 2130706431 [Aug 18 10:34:34] DEBUG[15430] rtp_engine.c: RTP instance '0x7f0cb00883d0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) ICE stopped [Aug 18 10:34:34] DEBUG[15430] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15430] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15430] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15430] res_rtp_asterisk.c: (0x7f0cb00883d0) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15430] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15430] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15430] chan_sip.c: SIP call-id changed from '5a9ed7924f0f61bc7d006fd0291f6640@127.0.1.1:5060' to '6966d7486050035f2ba5cf94424da3f8@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15430] stasis.c: Creating topic. name: channel:213228, detail: [Aug 18 10:34:34] DEBUG[15430] stasis.c: Topic 'channel:213228': 0x7f0cb011eee0 created [Aug 18 10:34:34] DEBUG[15430] stasis.c: Creating topic. name: cache:815/channel:213228, detail: [Aug 18 10:34:34] DEBUG[15430] stasis.c: Topic 'cache:815/channel:213228': 0x7f0cb006ba80 created [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE creating session 0.0.0.0:16544 (16544) [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE create [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #118 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #118)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116825@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK608439d6 Max-Forwards: 70 From: ;tag=as5d5338d2 To: Contact: Call-ID: 4ac20aa01240fb4f043474b45dbc159e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 479054008 479054008 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17982 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE add system candidates [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (2) INVITE - 5 [Aug 18 10:34:34] VERBOSE[15432] chan_sip.c: Audio is at 13638 [Aug 18 10:34:34] VERBOSE[15416] dial.c: Called zvonobot/79821116822 [Aug 18 10:34:34] DEBUG[13695] res_rtp_asterisk.c: (0x7f0ca800cb50) RTCP got report of 100 bytes from 178.62.121.41:16541 [Aug 18 10:34:34] DEBUG[15426] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116823@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a639eb Max-Forwards: 70 From: ;tag=as426458c9 To: Contact: Call-ID: 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1198383622 1198383622 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15426] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] VERBOSE[15432] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #11 (5) INVITE - 5 [Aug 18 10:34:34] WARNING[15106] app.c: No audio available on Recorder/ARI-00000047;1?? [Aug 18 10:34:34] VERBOSE[15106] app.c: User hung up [Aug 18 10:34:34] DEBUG[15106] res_stasis_recording.c: 1629282861.469: Recording complete [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #11)) [Aug 18 10:34:34] VERBOSE[15432] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE add candidate: 159.65.48.104:16544, 2130706431 [Aug 18 10:34:34] DEBUG[15106] channel.c: Channel 0x7f0ca4043bc0 'Recorder/ARI-00000047;1' hanging up. Refs: 2 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116884@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5b0f59d9 Max-Forwards: 70 From: ;tag=as293daefc To: Contact: Call-ID: 24be5aa55745a5980ce06ef42697de00@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116884@178.62.121.41", nonce="3e75e235", response="d1a471a4e25bb257da37004aaa39f357" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 320204973 320204974 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13406 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15320] channel.c: Channel 0x7f0cb0094430 'SIP/zvonobot-00000102' allocated [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:34] DEBUG[15320] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #24 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #24)) [Aug 18 10:34:34] DEBUG[15426] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15426] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE add candidate: 10.131.0.10:16544, 2130706431 [Aug 18 10:34:34] DEBUG[15426] rtp_engine.c: RTP instance '0x7f0c9c0b18f0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) ICE stopped [Aug 18 10:34:34] DEBUG[15426] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15426] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15426] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15426] res_rtp_asterisk.c: (0x7f0c9c0b18f0) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15426] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116828@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK162f95df Max-Forwards: 70 From: ;tag=as7b799277 To: Contact: Call-ID: 1170ea4763a17ffd27ea9e7569199057@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 670818967 670818967 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15460 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] VERBOSE[15432] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15370] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Aug 18 10:34:34] DEBUG[15426] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15426] chan_sip.c: SIP call-id changed from '203738434c61173b0bf7cf34436fa76f@127.0.1.1:5060' to '2fba28ef7d76d5c24317a1095e96a5bf@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15426] stasis.c: Creating topic. name: channel:213230, detail: [Aug 18 10:34:34] DEBUG[15426] stasis.c: Topic 'channel:213230': 0x7f0c9c038500 created [Aug 18 10:34:34] DEBUG[15426] stasis.c: Creating topic. name: cache:816/channel:213230, detail: [Aug 18 10:34:34] DEBUG[15426] stasis.c: Topic 'cache:816/channel:213230': 0x7f0c9c035560 created [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Initializing initreq for method INVITE - callid 32d69425685fae543fe99f5e243b59f6@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #110 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116817@178.62.121.41 SIP/2.0 [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: Allocating new SIP dialog for 6dd5426121d0d9b5247211d154e4e886@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15431] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0cac06a5e0' [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) RTP allocated port 13208 [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE creating session 0.0.0.0:13208 (13208) [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE create [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e1bdccb [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE add system candidates [Aug 18 10:34:34] DEBUG[15431] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15431] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE add candidate: 159.65.48.104:13208, 2130706431 [Aug 18 10:34:34] DEBUG[15431] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15431] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE add candidate: 10.131.0.10:13208, 2130706431 [Aug 18 10:34:34] DEBUG[15431] rtp_engine.c: RTP instance '0x7f0cac06a5e0' is setup and ready to go [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) ICE stopped [Aug 18 10:34:34] DEBUG[15431] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[15431] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[15431] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[15431] res_rtp_asterisk.c: (0x7f0cac06a5e0) RTCP setup on RTP instance [Aug 18 10:34:34] VERBOSE[15431] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #110)) [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 3 [ 52]: From: ;tag=as2448723a [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116844@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK65517f9d Max-Forwards: 70 From: ;tag=as7417feac To: Contact: Call-ID: 7766b4206901774644408a490f81d9d2@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 316590492 316590492 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14858 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15431] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15431] chan_sip.c: SIP call-id changed from '6dd5426121d0d9b5247211d154e4e886@127.0.1.1:5060' to '6219cbc20922c6e91ae84a6567c6801d@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15431] stasis.c: Creating topic. name: channel:213233, detail: [Aug 18 10:34:34] DEBUG[15431] stasis.c: Topic 'channel:213233': 0x7f0cac00f080 created [Aug 18 10:34:34] DEBUG[15431] stasis.c: Creating topic. name: cache:817/channel:213233, detail: [Aug 18 10:34:34] DEBUG[15431] stasis.c: Topic 'cache:817/channel:213233': 0x7f0cac091bb0 created [Aug 18 10:34:34] DEBUG[15433] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[15433] http.c: HTTP Request URI is /ari/channels/213003/snoop?app=calls_0&spy=in [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:34] DEBUG[15433] http.c: match request [ari/channels/213003/snoop] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15433] http.c: match request [ari/channels/213003/snoop] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15433] http.c: match request [ari/channels/213003/snoop] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:34] DEBUG[15433] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: Asked to create a SIP channel with formats: (slin192) [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: Allocating new SIP dialog for 47b144c24e4e126d0047caf41901ed03@127.0.1.1:5060 - INVITE (No RTP) [Aug 18 10:34:34] DEBUG[15428] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f0ca0023620' [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) RTP allocated port 10696 [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE creating session 0.0.0.0:10696 (10696) [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 6 [ 60]: Call-ID: 32d69425685fae543fe99f5e243b59f6@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE create [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Finding handler for channels/213003/snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE add system candidates [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:34 GMT [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] DEBUG[15428] netsock2.c: Splitting '159.65.48.104' into... [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Finding handler for 213003 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channels create: Didn't match 213003 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116818@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4f86ec Max-Forwards: 70 From: ;tag=as0c05fdf2 To: Contact: Call-ID: 359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 161411965 161411965 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18248 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] VERBOSE[15432] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116817@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e1bdccb Max-Forwards: 70 From: ;tag=as2448723a To: Contact: Call-ID: 32d69425685fae543fe99f5e243b59f6@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 982586535 982586535 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13638 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15428] netsock2.c: ...host '159.65.48.104' and port ''. [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE add candidate: 159.65.48.104:10696, 2130706431 [Aug 18 10:34:34] DEBUG[15428] netsock2.c: Splitting '10.131.0.10' into... [Aug 18 10:34:34] DEBUG[15428] netsock2.c: ...host '10.131.0.10' and port ''. [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE add candidate: 10.131.0.10:10696, 2130706431 [Aug 18 10:34:34] DEBUG[15428] rtp_engine.c: RTP instance '0x7f0ca0023620' is setup and ready to go [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) ICE stopped [Aug 18 10:34:34] DEBUG[15428] netsock2.c: Splitting 'GolosBetaAsterisk-01' into... [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #4 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #4)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116865@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK76dda519 Max-Forwards: 70 From: ;tag=as4bc9e76f To: Contact: Call-ID: 0332baa355a71ce92bc2431b256cb224@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116865@178.62.121.41", nonce="328f7607", response="03e2ed289df65658183229073b55646c" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 36230767 36230768 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16814 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #88 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #88)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116869@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2a0c0c7f Max-Forwards: 70 From: ;tag=as73a421e4 To: Contact: Call-ID: 1b33a3207d9d2cf55dca5557645f408b@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116869@178.62.121.41", nonce="59ecf5fc", response="092ee6197c12842bc052f0e289c8a55c" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1415909761 1415909762 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #139 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #139)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116842@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK61d82c2b Max-Forwards: 70 From: ;tag=as539445e1 To: Contact: Call-ID: 46be811217bc41126929634752a2647e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1223352312 1223352312 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11540 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #35 (5) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #35)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #5 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116868@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35098e23 Max-Forwards: 70 From: ;tag=as406d4539 To: Contact: Call-ID: 25018a477b0508ef13ab4fd67afa8e12@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116868@178.62.121.41", nonce="222035a3", response="df1e2eff61fa3e44aa0b11316f50a52f" Date: Wed, 18 Aug 2021 10:34:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1317398572 1317398573 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14666 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #46 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #46)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116859@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK0b84ac14 Max-Forwards: 70 From: ;tag=as0228d9c2 To: Contact: Call-ID: 447a50ba7810af590f60b91a27726642@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116859@178.62.121.41", nonce="4cb8ffaa", response="34d7073f228c83199f1d8a15ca6aeca9" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1263155085 1263155086 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19846 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ecf3464 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116851@178.62.121.41", nonce="71de2929", response="a192cb8c01ae51fd5bd60899bae739de" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876616 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #95 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #95)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116827@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK39a5b26d Max-Forwards: 70 From: ;tag=as568913af To: Contact: Call-ID: 45b0b22714aa16de060380717e05b99e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 962154366 962154366 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #2 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #2)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116867@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK04faa4b4 Max-Forwards: 70 From: ;tag=as7d725550 To: Contact: Call-ID: 3334457d359183f4783a1ff53f046a25@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116867@178.62.121.41", nonce="153bdf62", response="88c633d268e35d5ca7cd4960941d4e3b" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 2040706901 2040706902 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16168 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #112 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #112)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116860@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK26886e91 Max-Forwards: 70 From: ;tag=as17ad889a To: Contact: Call-ID: 2f459d660bd8cd56049142c300dbc344@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116860@178.62.121.41", nonce="09aacdf1", response="3c5bdea99ab5d5c94251dcfc28700249" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1003890280 1003890281 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 13594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channels externalMedia: Didn't match 213003 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Aug 18 10:34:34] DEBUG[15432] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15428] netsock2.c: ...host 'GolosBetaAsterisk-01' and port ''. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #87 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: No explicit handler found for 213003. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #87)) [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Finding handler for snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId continue: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId move: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId redirect: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId answer: Didn't match snoop [Aug 18 10:34:34] VERBOSE[15432] dial.c: Called zvonobot/79821116817 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId ring: Didn't match snoop [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId dtmf: Didn't match snoop [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116856@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK06fea4c6 Max-Forwards: 70 From: ;tag=as6b79f1a3 To: Contact: Call-ID: 08f54ecc1240fd0b3e4d9ec505600c79@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116856@178.62.121.41", nonce="4ecc103f", response="5984a684e9cf4d2ea38b7a835805bdb8" Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=root 96029194 96029195 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId mute: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId hold: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId moh: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId silence: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId play: Didn't match snoop [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15428] acl.c: Multiple addresses. Using the first only [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #123 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #123)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116854@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5532a489 Max-Forwards: 70 From: ;tag=as4b888052 To: Contact: Call-ID: 40bff8191ebc14ac3a4b50a51031a2bf@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116854@178.62.121.41", nonce="71b13d8e", response="1085dd2b15e8d0125cea2557194bb890" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1989090437 1989090438 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 17348 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId record: Didn't match snoop [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId variable: Didn't match snoop [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15433] res_ari.c: Checking channelId snoop: Explicit match with snoop [Aug 18 10:34:34] DEBUG[15428] res_rtp_asterisk.c: (0x7f0ca0023620) RTCP setup on RTP instance [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #89 (4) INVITE - 5 [Aug 18 10:34:34] DEBUG[15317] channel.c: Channel 0x7f0c98094020 'SIP/zvonobot-00000103' allocated [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: *** Our native formats are (alaw) [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: *** Joint capabilities are (nothing) [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: *** Our capabilities are (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: *** Our preferred formats from the incoming channel are (slin192) [Aug 18 10:34:34] DEBUG[15317] chan_sip.c: This channel will not be able to handle video. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #89)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #4 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116830@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK00cbebfd Max-Forwards: 70 From: ;tag=as39b021ce To: Contact: Call-ID: 0817506b3a45cdf526fc048c0750414e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1100223183 1100223183 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 16394 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Auto destroying SIP dialog '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #82 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #82)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116821@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK104fda02 Max-Forwards: 70 From: ;tag=as3a26bfb9 To: Contact: Call-ID: 315c846717ed98e20a6d06cb7f10d15e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1544986695 1544986695 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15514 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #55 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #55)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116863@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK5e6ce84c Max-Forwards: 70 From: ;tag=as48cc2656 To: Contact: Call-ID: 63fd4b59763be1fa2a5c41ff07805538@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116863@178.62.121.41", nonce="59bb839c", response="1881b128747b5422f1976fbb90797199" Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 118661387 118661388 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 10056 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: chan_sip: ast_sched_runq ran 22 all at once [Aug 18 10:34:34] DEBUG[15320] res_stasis.c: calls_0: Subscribing to 213220 [Aug 18 10:34:34] DEBUG[15320] stasis/app.c: Channel '213220' is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[15320] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:34] DEBUG[15320] http.c: HTTP closing session. Top level [Aug 18 10:34:34] VERBOSE[15428] netsock2.c: Using SIP RTP CoS mark 5 [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Outgoing Call for 79821116820 [Aug 18 10:34:34] DEBUG[15226] channel.c: Soft-Hanging (0x20) up channel 'UnicastRTP/127.0.0.1:50312-0x7f0c240eada0' [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: <--- SIP read from UDP:178.62.121.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8;received=159.65.48.104 From: ;tag=as0a05f417 To: ;tag=as029966c2 Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 CSeq: 102 INVITE Server: Asterisk PBX 16.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b16ead8" Content-Length: 0 <-------------> [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6e9dadf8;received=159.65.48.104 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 2 [ 52]: From: ;tag=as0a05f417 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 3 [ 50]: To: ;tag=as029966c2 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 4 [ 60]: Call-ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 16.18.0 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[15331] channel.c: Channel 0x7f0c280b5380 'Announcer/ARI-0000005a;1' destroying [Aug 18 10:34:34] DEBUG[15226] http.c: HTTP keeping session open. status_code:204 [Aug 18 10:34:34] DEBUG[15206] stasis/app.c: channel 'robot_212996': is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[15206] channel.c: Channel 0x7f0c241243f0 'UnicastRTP/127.0.0.1:50312-0x7f0c240eada0' hanging up. Refs: 2 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 464 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 464 [Aug 18 10:34:34] DEBUG[15226] http.c: HTTP closing session. Top level [Aug 18 10:34:34] DEBUG[15324] bridge_channel.c: Setting 0x7f0c280f4d10(Announcer/ARI-0000005a;2) state from:0 to:1 [Aug 18 10:34:34] DEBUG[15436] http.c: HTTP opening session. Top level [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[15436] http.c: HTTP Request URI is /ari/channels/212996 [Aug 18 10:34:34] DEBUG[15436] http.c: match request [ari/channels/212996] with handler [httpstatus] len 10 [Aug 18 10:34:34] DEBUG[15436] http.c: match request [ari/channels/212996] with handler [phoneprov] len 9 [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 18 10:34:34] DEBUG[15436] http.c: match request [ari/channels/212996] with handler [ari] len 3 [Aug 18 10:34:34] DEBUG[15436] http.c: Match made with [ari] [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Finding handler for channels/212996 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Finding handler for channels [Aug 18 10:34:34] DEBUG[15428] acl.c: For destination '178.62.121.41', our source address is '159.65.48.104'. [Aug 18 10:34:34] DEBUG[15331] stasis.c: Destroying topic. name: cache:712/channel:1629282870.621, detail: [Aug 18 10:34:34] DEBUG[15331] stasis.c: Topic 'cache:712/channel:1629282870.621': 0x7f0c280efa90 destroyed [Aug 18 10:34:34] DEBUG[15331] stasis.c: Destroying topic. name: channel:1629282870.621, detail: [Aug 18 10:34:34] DEBUG[15331] stasis.c: Topic 'channel:1629282870.621': 0x7f0c280a4a20 destroyed [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking ari endpoints: Didn't match channels [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] DEBUG[15317] res_stasis.c: calls_0: Subscribing to 213221 [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking ari events: Didn't match channels [Aug 18 10:34:34] VERBOSE[15434] chan_sip.c: Audio is at 15326 [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking ari applications: Didn't match channels [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking ari recordings: Didn't match channels [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking ari channels: Explicit match with channels [Aug 18 10:34:34] VERBOSE[15434] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Finding handler for 212996 [Aug 18 10:34:34] DEBUG[15324] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: pulling 0x7f0c280f4d10(Announcer/ARI-0000005a;2) [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking channels create: Didn't match 212996 [Aug 18 10:34:34] VERBOSE[15434] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking channels channelId: Matched wildcard. [Aug 18 10:34:34] VERBOSE[15324] bridge_channel.c: Channel Announcer/ARI-0000005a;2 left 'simple_bridge' stasis-bridge <9cefb3ad-33ea-4a52-96a1-42b677d6802c> [Aug 18 10:34:34] DEBUG[15324] bridge_channel.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c: 0x7f0c280f4d10(Announcer/ARI-0000005a;2) is leaving simple_bridge technology [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: Setting AST_TRANSPORT_UDP with address 159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: Setting NAT on RTP to Off [Aug 18 10:34:34] DEBUG[15428] chan_sip.c: SIP call-id changed from '47b144c24e4e126d0047caf41901ed03@127.0.1.1:5060' to '230c46e965c5b1152a636e1a2cd333f7@159.65.48.104:5060' [Aug 18 10:34:34] DEBUG[15428] stasis.c: Creating topic. name: channel:213231, detail: [Aug 18 10:34:34] DEBUG[15428] stasis.c: Topic 'channel:213231': 0x7f0ca00ee7b0 created [Aug 18 10:34:34] DEBUG[15428] stasis.c: Creating topic. name: cache:818/channel:213231, detail: [Aug 18 10:34:34] DEBUG[15428] stasis.c: Topic 'cache:818/channel:213231': 0x7f0ca005cce0 created [Aug 18 10:34:34] DEBUG[15436] res_ari.c: Checking channels externalMedia: Didn't match 212996 [Aug 18 10:34:34] VERBOSE[15434] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[15436] res_ari.c: No explicit handler found for 212996. Using wildcard channelId. [Aug 18 10:34:34] DEBUG[15317] stasis/app.c: Channel '213221' is 1 interested in calls_0 [Aug 18 10:34:34] DEBUG[15324] bridge_native_rtp.c: Bridge '9cefb3ad-33ea-4a52-96a1-42b677d6802c' can not use native RTP bridge as two channels are required [Aug 18 10:34:34] DEBUG[15324] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Aug 18 10:34:34] DEBUG[15324] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Aug 18 10:34:34] DEBUG[15324] bridge.c: Bridge technology softmix does not have any capabilities we want. [Aug 18 10:34:34] DEBUG[15324] bridge.c: Chose bridge technology simple_bridge [Aug 18 10:34:34] DEBUG[15324] bridge.c: Bridge 9cefb3ad-33ea-4a52-96a1-42b677d6802c is already using the new technology. [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b16ead8" [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Outgoing Call for 79821116819 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Updating call counter for outgoing call [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: False Text flag: False [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: ** Our prefcodec: (slin192) [Aug 18 10:34:34] VERBOSE[15437] chan_sip.c: Audio is at 12594 [Aug 18 10:34:34] VERBOSE[15437] chan_sip.c: Adding codec alaw to SDP [Aug 18 10:34:34] DEBUG[15324] channel.c: Channel 0x7f0c2815a320 'Announcer/ARI-0000005a;2' hanging up. Refs: 2 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15317] http.c: HTTP keeping session open. status_code:200 [Aug 18 10:34:34] VERBOSE[15437] chan_sip.c: Adding codec ulaw to SDP [Aug 18 10:34:34] VERBOSE[15437] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: -- Done with adding codecs to SDP [Aug 18 10:34:34] DEBUG[15317] http.c: HTTP closing session. Top level [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Initializing initreq for method INVITE - callid 7e9a053f122d7f9206b4bd144cc6c1b5@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw) [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116820@178.62.121.41 SIP/2.0 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Initializing initreq for method INVITE - callid 2aede3cb384162731ba8d58b471d2348@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35ea34d6 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 0 [ 44]: INVITE sip:79821116819@178.62.121.41 SIP/2.0 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: --- (11 headers 0 lines) --- [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6760151f [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: = Looking for Call ID: 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 (Checking To) --From tag as0a05f417 --To-tag as029966c2 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 3676e98e6d3120834b4bd7fa24e0c17e@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Invalid SIP message - rejected , no callid, len 529 [Aug 18 10:34:34] WARNING[20585] chan_sip.c: Retransmission timeout reached on transmission 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6504ms with no response [Aug 18 10:34:34] WARNING[20585] chan_sip.c: Hanging up call 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6712e58b01f7a22d5e2995561594c93d@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 3 [ 52]: From: ;tag=as7a876837 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 3 [ 52]: From: ;tag=as13715cb5 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 4 [ 35]: To: [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 5 [ 45]: Contact: [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #90 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #90)) [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 6 [ 60]: Call-ID: 2aede3cb384162731ba8d58b471d2348@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 6 [ 60]: Call-ID: 7e9a053f122d7f9206b4bd144cc6c1b5@159.65.48.104:5060 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116824@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK69fe0e7d Max-Forwards: 70 From: ;tag=as6f815416 To: Contact: Call-ID: 2d3f1e35624a05e23ccb1716007dba6e@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 937031862 937031862 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14390 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15060] channel.c: Channel 0x7f0cb40a3910 'SIP/zvonobot-000000d4' hanging up. Refs: 2 [Aug 18 10:34:34] DEBUG[15326] app.c: One waitfor failed, trying another [Aug 18 10:34:34] DEBUG[20545] cdr.c: Finalized CDR for SIP/zvonobot-000000d4 - start 1629282868.005987 answer 0.000000 end 1629282874.736550 dur 6.730 bill 1629282874.736 dispo NO ANSWER [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:34 GMT [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 16.20.0 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 9 [ 35]: Date: Wed, 18 Aug 2021 10:34:34 GMT [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 542 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 542 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #15 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:34] VERBOSE[15437] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116819@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK6760151f Max-Forwards: 70 From: ;tag=as13715cb5 To: Contact: Call-ID: 2aede3cb384162731ba8d58b471d2348@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1549793106 1549793106 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 12594 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #79 [Aug 18 10:34:34] DEBUG[15437] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] VERBOSE[15437] dial.c: Called zvonobot/79821116819 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #15)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116826@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK7881358b Max-Forwards: 70 From: ;tag=as5531004e To: Contact: Call-ID: 01e37a3836a4dad449f32b917ae97b06@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 715600489 715600489 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 18 10:34:34] VERBOSE[15434] chan_sip.c: Reliably Transmitting (no NAT) to 178.62.121.41:5060: INVITE sip:79821116820@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK35ea34d6 Max-Forwards: 70 From: ;tag=as7a876837 To: Contact: Call-ID: 7e9a053f122d7f9206b4bd144cc6c1b5@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 502536743 502536743 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 15326 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #131 [Aug 18 10:34:34] DEBUG[15434] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #16 (1) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #16)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #1 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116822@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c492a07 Max-Forwards: 70 From: ;tag=as1abebfe3 To: Contact: Call-ID: 509f8e0e3acc6f0a7d623d30256c2b1f@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 812289418 812289418 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 14728 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] VERBOSE[15434] dial.c: Called zvonobot/79821116820 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket string of length 534 [Aug 18 10:34:34] DEBUG[20620] res_http_websocket.c: Writing websocket text frame, length 534 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #81 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116818@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK3c4f86ec Max-Forwards: 70 From: ;tag=as0c05fdf2 To: Contact: Call-ID: 359a7f9a00dfc5857c7788f171bd4e85@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 161411965 161411965 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 18248 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #34 (2) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #34)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #2 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116851@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK2ecf3464 Max-Forwards: 70 From: ;tag=as5f8e1c47 To: Contact: Call-ID: 21879c63575c85bf5a3a7b4a628aadfb@159.65.48.104:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.20.0 Authorization: Digest username="zvonobot", realm="asterisk", algorithm=MD5, uri="sip:79821116851@178.62.121.41", nonce="71de2929", response="a192cb8c01ae51fd5bd60899bae739de" Date: Wed, 18 Aug 2021 10:34:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 264 v=0 o=root 388876615 388876616 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 19388 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: SIP TIMER: Rescheduling retransmission #114 (3) INVITE - 5 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #114)) [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Retransmitting #3 (no NAT) to 178.62.121.41:5060: INVITE sip:79821116823@178.62.121.41 SIP/2.0 Via: SIP/2.0/UDP 159.65.48.104:5060;branch=z9hG4bK02a639eb Max-Forwards: 70 From: ;tag=as426458c9 To: Contact: Call-ID: 7c5feaaf4e269ddd7d1e81081f9677b9@159.65.48.104:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.20.0 Date: Wed, 18 Aug 2021 10:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 266 v=0 o=root 1198383622 1198383622 IN IP4 159.65.48.104 s=Asterisk PBX 16.20.0 c=IN IP4 159.65.48.104 t=0 0 m=audio 11192 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 178.62.121.41:5060 [Aug 18 10:34:34] DEBUG[20585] chan_sip.c: Destroying SIP dialog 3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060 [Aug 18 10:34:34] VERBOSE[20585] chan_sip.c: Really destroying SIP dialog '3915c4f87b526f5f15ec3731599ffbfe@159.65.48.104:5060' Method: BYE [Aug 18 10:34:34] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS stop [Aug 18 10:34:34] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:34] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) DTLS srtp - stopped timeout timer' [Aug 18 10:34:34] DEBUG[20585] res_rtp_asterisk.c: (0x7f0c24007240) ICE RTP transport deallocating [Aug 18 10:34:34] DEBUG[20585] rtp_engine.c: Destroyed RTP instance '0x7f0c24007240' [Aug 18 10:34:34] DEBUG[15202] channel.c: Channel 0x7f0c08080020 'Recorder/ARI-0000004c;1' destroying [Aug 18 10:34:34] DEBUG[12874] chan_sip.c: Hangup call SIP/zvonobot-00000002, SIP callid 261fc34e44833caa0ca1c43617b0fe86@159.65.48.104:5060 [Aug 18 10:34:34] DEBUG[12874] res_rtp_asterisk.c: (0x7f0c1000e000) DTLS srtp - stopped timeout timer'